<?xml version="1.0" encoding="UTF-8"?>
<otherdata xmlns="http://linux.duke.edu/metadata/other" packages="668">
<package pkgid="5acd6ced560f089fd36f66fea936849d87c8fb80c646f5394d9b9a687bed2d73" name="DHCProbe" arch="src">
  <version epoch="0" ver="0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.1-1" date="1544184000">- First version of spec</changelog>
</package>
<package pkgid="f80a00128c0f7bdbe678f401906329c1b13298a578e319b0bb817d842940ec5d" name="SDL2" arch="src">
  <version epoch="0" ver="2.0.10" rel="1.el7_9"/>
  <changelog author="Miro Hron&#x10D;ok &lt;mhroncok@redhat.com&gt; - 2.0.8-2" date="1520164800">- Disable altivec on ppc64le (RHBZ #1551338)</changelog>
  <changelog author="David Abdurachmanov &lt;david.abdurachmanov@gmail.com&gt; - 2.0.8-3" date="1522411200">- Add riscv64 to SDL_config.h</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.0.8-4" date="1523448000">- enable video-kmsdrm</changelog>
  <changelog author="Adam Jackson &lt;ajax@redhat.com&gt; - 2.0.8-5" date="1526904000">- Backport a crash/hang fix from 2.0.9 (#1580541)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.0.8-6" date="1531396800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.0.9-1" date="1541160000">- update to 2.0.9</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.0.9-2" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.0.9-3" date="1550232000">- use khrplatform defines, not ptrdiff_t</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.0.9-4" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Pete Walter &lt;pwalter@fedoraproject.org&gt; - 2.0.10-1" date="1564142400">- Update to 2.0.10</changelog>
</package>
<package pkgid="561e3dc483e00f9bf346c6e614ea3d68bf9b30451b9f217f422ba1ca02224b29" name="SDL2" arch="src">
  <version epoch="0" ver="2.0.8" rel="5.el7_9"/>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 2.0.6-4" date="1508414400">- Fully fix last overflow</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 2.0.7-1" date="1508846400">- Update to 2.0.7</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 2.0.7-2" date="1509883200">- Fix IBus</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 2.0.7-3" date="1517400000">- Switch to %ldconfig_scriptlets</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.0.7-4" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 2.0.8-1" date="1520164800">- Update to 2.0.8</changelog>
  <changelog author="Miro Hron&#x10D;ok &lt;mhroncok@redhat.com&gt; - 2.0.8-2" date="1520164801">- Disable altivec on ppc64le (RHBZ #1551338)</changelog>
  <changelog author="David Abdurachmanov &lt;david.abdurachmanov@gmail.com&gt; - 2.0.8-3" date="1522411200">- Add riscv64 to SDL_config.h</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.0.8-4" date="1523448000">- enable video-kmsdrm</changelog>
  <changelog author="Adam Jackson &lt;ajax@redhat.com&gt; - 2.0.8-5" date="1526904000">- Backport a crash/hang fix from 2.0.9 (#1580541)</changelog>
</package>
<package pkgid="e334db69e901298ac7c1c4866c3e36eae5f580e91696cbdb3a8d5618cb4f9a15" name="SDL_image" arch="src">
  <version epoch="0" ver="1.2.12" rel="11.el7_9"/>
  <changelog author="Jon Ciesla &lt;limburgher@gmail.com&gt; - 1.2.12-1" date="1328011200">- New upstream.
- libpng15 patch upstreamed.</changelog>
  <changelog author="Jon Ciesla &lt;limburgher@gmail.com&gt; - 1.2.12-2" date="1336392000">- Rebuild for new libtiff.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.2.12-3" date="1342612800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Adam Tkac &lt;atkac redhat com&gt; - 1.2.12-4" date="1356091200">- rebuild against new libjpeg</changelog>
  <changelog author="Adam Tkac &lt;atkac redhat com&gt; - 1.2.12-5" date="1358510400">- rebuild due to "jpeg8-ABI" feature drop</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.2.12-6" date="1360756800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.2.12-7" date="1375444800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.2.12-8" date="1402056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.2.12-9" date="1408104000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Moez Roy &lt;moez.roy@gmail.com&gt; - 1.2.12-11" date="1414065600">- Add global harderning flags</changelog>
</package>
<package pkgid="485a1dedab85668594632ce6d077f2e328cb3833941bb42e3da9432c52da7532" name="a52dec" arch="src">
  <version epoch="0" ver="0.7.4" rel="27.el7_9"/>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.7.4-17" date="1362312000">- Mass rebuilt for Fedora 19 Features</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.7.4-18" date="1362916800">- Add silence patch as we don't built with DJBFFT enabled</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.7.4-19" date="1409400000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.7.4-20" date="1471348800">- Clean spec, with Vascom, rfbz #4193, add license tag</changelog>
  <changelog author="Nicolas Chauvet &lt;nicolas.chauvet@kwizart.fr&gt; - 0.7.4-21" date="1471953600">- Fix macro in comment</changelog>
  <changelog author="RPM Fusion Release Engineering &lt;kwizart@rpmfusion.org&gt; - 0.7.4-22" date="1489838400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.7.4-24" date="1490011200">- Multilibs support - rhbz#1433758
- Simplify description
- Convert AUTHORS to UTF-8
- Drop Groups
- Add Obsoletes/Provides for a52dec-devel</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.7.4-25" date="1490097600">- Fixup Obsoletes/Provides for the devel
- Use sed instead of perl to avoid a build dependency</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.7.4-26" date="1490875200">- Fix multilibs transition</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.7.4-27" date="1491307200">- Fix others 3rd part repos</changelog>
</package>
<package pkgid="8a16861662ed72e4d7855e37cf224b4b6c15943461459ad47892c6408d30e3b2" name="adobe-source-code-pro-fonts" arch="src">
  <version epoch="0" ver="2.010.1.030" rel="1.el7_9"/>
  <changelog author="Tobias Florek me@ibotty.net - 1.010-1" date="1349179200">- new upstream release</changelog>
  <changelog author="Tobias Florek me@ibotty.net - 1.010-2" date="1353240000">- do not clean buildroot on install and clean</changelog>
  <changelog author="Tobias Florek me@ibotty.net - 1.013-1" date="1354881600">- new upstream release</changelog>
  <changelog author="Tobias Florek me@ibotty.net - 1.017-1" date="1358424000">- new upstream release</changelog>
  <changelog author="Tobias Florek me@ibotty.net - 1.017-2" date="1358769600">- remove now empty clean section</changelog>
  <changelog author="Matt Rose mattrose@folkwolf.net - 1.017-3" date="1379073600">- rebuilt on fedora 19</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.017-4" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Richard Hughes &lt;richard@hughsie.com&gt; - 1.017-5" date="1413374400">- Add a MetaInfo file for the software center; this is a font we want to show.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.017-6" date="1434456000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Jamie Nguyen &lt;jamielinux@fedoraproject.org&gt; - 2.010.1.030-1" date="1446811200">- update to 2.010 (roman) and 1.030 (italic)
- update URL
- use %license macro</changelog>
</package>
<package pkgid="96c0f8da1e8ee974f321758777bddd959778fd86ddbe3dff54542c3cd94c1f87" name="adobe-source-sans-pro-fonts" arch="src">
  <version epoch="0" ver="2.020" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.034-2" date="1360756800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Alexis Lameire &lt;alexisis-pristontale@hotmail.com&gt; - 1.050-1" date="1367236800">- update to 1.050-1 upstream release</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.050-2" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.050-3" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Richard Hughes &lt;richard@hughsie.com&gt; - 1.050-4" date="1413374400">- Add a MetaInfo file for the software center; this is a font we want to show.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.050-5" date="1434456000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.050-6" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.050-7" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.050-8" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Mohamed El Morabity &lt;melmorabity@fedoraproject.org&gt; - 2.020-1" date="1508241600">- Update to 2.020
- Update upstream URL
- Comply with latest packaging guidelines</changelog>
</package>
<package pkgid="b5c080bea158c4df488213ded406adfadf3ae5b2d34105c724b43f986c5d2d97" name="ag800lcd" arch="src">
  <version epoch="0" ver="2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.1-1" date="1504353600">- First version of this spec file</changelog>
</package>
<package pkgid="e2fc78c76bb0e94088e7da4ff9898a07f2776c57b8b975f4d71f3b3a7916e9fb" name="amiws" arch="src">
  <version epoch="0" ver="2.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
</package>
<package pkgid="e700591a8e9754080c0fb8e4b7b57bffef9513aff317ac201ad320f7b4cfa1d4" name="amiws" arch="src">
  <version epoch="0" ver="2.3.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1673697600">- Updated to release 2.3.1</changelog>
</package>
<package pkgid="30e9c82a7c809e25bc16eb60f4aa7c09d8c41527b938a7c629ac16b563074549" name="amiws" arch="src">
  <version epoch="0" ver="2.3.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1508068800">- Initial build</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1673697600">- Updated to release 2.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.2-1" date="1678363200">- Updated to release 2.3.2</changelog>
</package>
<package pkgid="334614642d607bc7346c8bbfd8539f1fcc750f29a4fb7f50b5c3c93d7043301c" name="arts" arch="src">
  <version epoch="8" ver="1.5.10" rel="26.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-17" date="1342612800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Rex Dieter &lt;rdieter@fedoraproject.org&gt; 1.5.10-18" date="1349956800">- rebuild (audiofile)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-19" date="1360756800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Than Ngo &lt;than@redhat.com&gt; - 1.5.10-20" date="1362657600">- fix FTBFS in rawhide</changelog>
  <changelog author="Kevin Kofler &lt;Kevin@tigcc.ticalc.org&gt; - 1.5.10-21" date="1362830400">- unify KDE 3 autotools fixes between packages</changelog>
  <changelog author="Kevin Kofler &lt;Kevin@tigcc.ticalc.org&gt; - 1.5.10-22" date="1364299200">- use automake --force-missing to get aarch64 support (#925029/#925627)
- also use automake --copy (the default is symlinking)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-23" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-24" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Rex Dieter &lt;rdieter@fedoraproject.org&gt; 8:1.5.10-25" date="1407931200">- kill rpaths harder</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 8:1.5.10-26" date="1408104000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
</package>
<package pkgid="87bb89d7fa49b23549f0b4839a0c0b4a7ae92773f7def95dcd6929434ebb6826" name="asio" arch="src">
  <version epoch="0" ver="1.10.8" rel="1.el7_9"/>
  <changelog author="David Tardon &lt;dtardon@redhat.com&gt; - 1.10.4-5" date="1437566400">- rebuild for Boost 1.58</changelog>
  <changelog author="Fabio Alessandro Locati &lt;fale@fedoraproject.org&gt; -1.10.6-1" date="1437912000">- Update to 1.10.6 version</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.10.6-2" date="1438171200">- Rebuilt for https://fedoraproject.org/wiki/Changes/F23Boost159</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.10.6-3" date="1440676800">- Rebuilt for Boost 1.59</changelog>
  <changelog author="Fabio Alessandro Locati &lt;fale@fedoraproject.org&gt; - 1.10.6-4" date="1451736000">- Move from define to global</changelog>
  <changelog author="Fabio Alessandro Locati &lt;fale@fedoraproject.org&gt; - 1.10.6-5" date="1451736001">- Remove useless pieces of the spec
- Conform to more recent SPEC style
- Fix date in changelog that was giving warnings</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.10.6-6" date="1452772800">- Rebuilt for Boost 1.60</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.10.6-7" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Fabio Alessandro Locati &lt;fale@fedoraproject.org&gt; - 1.10.7-1" date="1473768000">- Update to 1.10.7</changelog>
  <changelog author="Fabio Alessandro Locati &lt;fale@fedoraproject.org&gt; - 1.10.8-1" date="1481457600">- Update to 1.10.8</changelog>
</package>
<package pkgid="b5c90b044728abb99db7a26f4227f656b17ff827dc5b63fb1d8d0b571948e23e" name="asterisk" arch="src">
  <version epoch="0" ver="11.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28-cert3, 11.6-cert8,
- 1.8.32.1, 11.14.1, 12.7.1, and 13.0.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2014-012: Unauthorized access in the presence of ACLs with mixed IP
-   address families
-
-   Many modules in Asterisk that service incoming IP traffic have ACL options
-   ("permit" and "deny") that can be used to whitelist or blacklist address
-   ranges. A bug has been discovered where the address family of incoming
-   packets is only compared to the IP address family of the first entry in the
-   list of access control rules. If the source IP address for an incoming
-   packet is not of the same address as the first ACL entry, that packet
-   bypasses all ACL rules.
-
- * AST-2014-018: Permission Escalation through DB dialplan function
-
-   The DB dialplan function when executed from an external protocol, such as AMI,
-   could result in a privilege escalation. Users with a lower class authorization
-   in AMI can access the internal Asterisk database without the required SYSTEM
-   class authorization.
-
- In addition, the release of 11.6-cert8 and 11.14.1 resolves the following
- security vulnerability:
-
- * AST-2014-014: High call load with ConfBridge can result in resource exhaustion
-
-   The ConfBridge application uses an internal bridging API to implement
-   conference bridges. This internal API uses a state model for channels within
-   the conference bridge and transitions between states as different things
-   occur. Unload load it is possible for some state transitions to be delayed
-   causing the channel to transition from being hung up to waiting for media. As
-   the channel has been hung up remotely no further media will arrive and the
-   channel will stay within ConfBridge indefinitely.
-
- In addition, the release of 11.6-cert8, 11.14.1, 12.7.1, and 13.0.1 resolves
- the following security vulnerability:
-
- * AST-2014-017: Permission Escalation via ConfBridge dialplan function and
-                 AMI ConfbridgeStartRecord Action
-
-   The CONFBRIDGE dialplan function when executed from an external protocol (such
-   as AMI) can result in a privilege escalation as certain options within that
-   function can affect the underlying system. Additionally, the AMI
-   ConfbridgeStartRecord action has options that would allow modification of the
-   underlying system, and does not require SYSTEM class authorization in AMI.
-
- Finally, the release of 12.7.1 and 13.0.1 resolves the following security
- vulnerabilities:
-
- * AST-2014-013: Unauthorized access in the presence of ACLs in the PJSIP stack
-
-   The Asterisk module res_pjsip provides the ability to configure ACLs that may
-   be used to reject SIP requests from various hosts. However, the module
-   currently fails to create and apply the ACLs defined in its configuration
-   file on initial module load.
-
- * AST-2014-015: Remote crash vulnerability in PJSIP channel driver
-
-   The chan_pjsip channel driver uses a queue approach for relating to SIP
-   sessions. There exists a race condition where actions may be queued to answer
-   a session or send ringing after a SIP session has been terminated using a
-   CANCEL request. The code will incorrectly assume that the SIP session is still
-   active and attempt to send the SIP response. The PJSIP library does not
-   expect the SIP session to be in the disconnected state when sending the
-   response and asserts.
-
- * AST-2014-016: Remote crash vulnerability in PJSIP channel driver
-
-   When handling an INVITE with Replaces message the res_pjsip_refer module
-   incorrectly assumes that it will be operating on a channel that has just been
-   created. If the INVITE with Replaces message is sent in-dialog after a session
-   has been established this assumption will be incorrect. The res_pjsip_refer
-   module will then hang up a channel that is actually owned by another thread.
-   When this other thread attempts to use the just hung up channel it will end up
-   using a freed channel which will likely result in a crash.
-
- For more information about the details of these vulnerabilities, please read
- security advisories AST-2014-012, AST-2014-013, AST-2014-014, AST-2014-015,
- AST-2014-016, AST-2014-017, and AST-2014-018, which were released at the
- same time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert8
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-012.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-013.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-014.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-015.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-016.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-017.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2014-018.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532802">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532803">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0a471da66b2dcf83823ef8d64ab1bd1b6328d3d8f9991ca2a6a18fff0a11e36d" name="asterisk" arch="src">
  <version epoch="0" ver="11.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.14.2-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and Asterisk 11, 12, and 13. The available security releases are
- released as versions 11.6-cert9, 11.14.2, 12.7.2, and 13.0.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2014-019: Remote Crash Vulnerability in WebSocket Server
-
-   When handling a WebSocket frame the res_http_websocket module dynamically
-   changes the size of the memory used to allow the provided payload to fit. If a
-   payload length of zero was received the code would incorrectly attempt to
-   resize to zero. This operation would succeed and end up freeing the memory but
-   be treated as a failure. When the session was subsequently torn down this
-   memory would get freed yet again causing a crash.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2014-019, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert9
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.14.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.7.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.0.2
-
- The security advisory is available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2014-019.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532801">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532802">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453032000">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d4bb0776a46fc67ccec750f545378ea3cdaf830b7e07a1889dbeae8ddb392d0f" name="asterisk" arch="src">
  <version epoch="0" ver="11.21.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.0-1" date="1422532800">- The Asterisk Development Team has announced the release of Asterisk 11.15.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-20127 - [Regression] Config.c config_text_file_load()
-       unescapes semicolons ("\;" -&gt; ";") turning them into comments
-       (corruption) on rewrite of a config file (Reported by George
-       Joseph)
-  * ASTERISK-24307 - Unintentional memory retention in stringfields
-       (Reported by Etienne Lessard)
-  * ASTERISK-24492 - main/file.c: ast_filestream sometimes causes
-       extra calls to ast_module_unref (Reported by Corey Farrell)
-  * ASTERISK-24504 - chan_console: Fix reference leaks to pvt
-       (Reported by Corey Farrell)
-  * ASTERISK-24468 - Incoming UCS2 encoded SMS truncated if SMS
-       length exceeds 50 (roughly) national symbols (Reported by
-       Dmitriy Bubnov)
-  * ASTERISK-24500 - Regression introduced in chan_mgcp by SVN
-       revision r227276 (Reported by Xavier Hienne)
-  * ASTERISK-20402 - Unable to cancel (features.conf) attended
-       transfer (Reported by Matt Riddell)
-  * ASTERISK-24505 - manager: http connections leak references
-       (Reported by Corey Farrell)
-  * ASTERISK-24502 - Build fails when dev-mode, dont optimize and
-       coverage are enabled (Reported by Corey Farrell)
-  * ASTERISK-24444 - PBX: Crash when generating extension for
-       pattern matching hint (Reported by Leandro Dardini)
-  * ASTERISK-24522 - ConfBridge: delay occurs between kicking all
-       endmarked users when last marked user leaves (Reported by Matt
-       Jordan)
-  * ASTERISK-15242 - transmit_refer leaks sip_refer structures
-       (Reported by David Woolley)
-  * ASTERISK-24440 - Call leak in Confbridge (Reported by Ben Klang)
-  * ASTERISK-24469 - Security Vulnerability: Mixed IPv4/IPv6 ACLs
-       allow blocked addresses through (Reported by Matt Jordan)
-  * ASTERISK-24516 - [patch]Asterisk segfaults when playing back
-       voicemail under high concurrency with an IMAP backend (Reported
-       by David Duncan Ross Palmer)
-  * ASTERISK-24572 - [patch]App_meetme is loaded without its
-       defaults when the configuration file is missing (Reported by
-       Nuno Borges)
-  * ASTERISK-24573 - [patch]Out of sync conversation recording when
-       divided in multiple recordings (Reported by Nuno Borges)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24283 - [patch]Microseconds precision in the eventtime
-       column in the cel_odbc module (Reported by Etienne Lessard)
-  * ASTERISK-24530 - [patch] app_record stripping 1/4 second from
-       recordings (Reported by Ben Smithurst)
-  * ASTERISK-24577 - Speed up loopback switches by avoiding unneeded
-       lookups (Reported by Birger "WIMPy" Harzenetter)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532801">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="abe0cca84f803a4e84bf88f2a79bbdacd453441e5a2b82a00c12f1fcc8f1af93" name="asterisk" arch="src">
  <version epoch="0" ver="11.21.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.15.1-1" date="1422532800">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28 and 11.6 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-4, 1.8.32.2, 11.6-cert10,
- 11.15.1, 12.8.1, and 13.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2015-001: File descriptor leak when incompatible codecs are offered
-
-                 Asterisk may be configured to only allow specific audio or
-                 video codecs to be used when communicating with a
-                 particular endpoint. When an endpoint sends an SDP offer
-                 that only lists codecs not allowed by Asterisk, the offer
-                 is rejected. However, in this case, RTP ports that are
-                 allocated in the process are not reclaimed.
-
-                 This issue only affects the PJSIP channel driver in
-                 Asterisk. Users of the chan_sip channel driver are not
-                 affected.
-
- * AST-2015-002: Mitigation for libcURL HTTP request injection vulnerability
-
-                 CVE-2014-8150 reported an HTTP request injection
-                 vulnerability in libcURL. Asterisk uses libcURL in its
-                 func_curl.so module (the CURL() dialplan function), as well
-                 as its res_config_curl.so (cURL realtime backend) modules.
-
-                 Since Asterisk may be configured to allow for user-supplied
-                 URLs to be passed to libcURL, it is possible that an
-                 attacker could use Asterisk as an attack vector to inject
-                 unauthorized HTTP requests if the version of libcURL
-                 installed on the Asterisk server is affected by
-                 CVE-2014-8150.
-
- For more information about the details of these vulnerabilities, please read
- security advisory AST-2015-001 and AST-2015-002, which were released at the same
- time as this announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert10
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.15.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.1.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2015-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2015-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
</package>
<package pkgid="6e455e9f54fff0313e263c71b74fbb30c5b882bcf4c423bbb211c88d0dec7d21" name="asterisk" arch="src">
  <version epoch="0" ver="11.21.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.16.0-1" date="1426593600">- The Asterisk Development Team has announced the release of Asterisk 11.16.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- * ASTERISK-24472 - Asterisk Crash in OpenSSL when calling over WSS
-      from JSSIP (Reported by Badalian Vyacheslav)
- * ASTERISK-24614 - Deadlock when DEBUG_THREADS compiler flag
-      enabled (Reported by Richard Mudgett)
- * ASTERISK-24449 - Reinvite for T.38 UDPTL fails if SRTP is
-      enabled (Reported by Andreas Steinmetz)
- * ASTERISK-24619 - [patch]Gcc 4.10 fixes in r413589 (1.8) wrongly
-      casts char to unsigned int (Reported by Walter Doekes)
- * ASTERISK-24337 - Spammy DEBUG message needs to be at a higher
-      level - 'Remote address is null, most likely RTP has been
-      stopped' (Reported by Rusty Newton)
- * ASTERISK-23733 - 'reload acl' fails if acl.conf is not present
-      on startup (Reported by Richard Kenner)
- * ASTERISK-24628 - [patch] chan_sip - CANCEL is sent to wrong
-      destination when 'sendrpid=yes' (in proxy environment) (Reported
-      by Karsten Wemheuer)
- * ASTERISK-24672 - [PATCH] Memory leak in func_curl CURLOPT
-      (Reported by Kristian Høgh)
- * ASTERISK-20744 - [patch] Security event logging does not work
-      over syslog (Reported by Michael Keuter)
- * ASTERISK-23850 - Park Application does not respect Return
-      Context Priority (Reported by Andrew Nagy)
- * ASTERISK-23991 - [patch]asterisk.pc file contains a small error
-      in the CFlags returned (Reported by Diederik de Groot)
- * ASTERISK-24288 - [patch] - ODBC usage with app_voicemail -
-      voicemail is not deleted after review, hangup (Reported by LEI
-      FU)
- * ASTERISK-24048 - [patch] contrib/scripts/install_prereq selects
-      32-bit packages on 64-bit hosts (Reported by Ben Klang)
- * ASTERISK-24709 - [patch] msg_create_from_file used by MixMonitor
-      m() option does not queue an MWI event (Reported by Gareth
-      Palmer)
- * ASTERISK-24355 - [patch] chan_sip realtime uses case sensitive
-      column comparison for 'defaultuser' (Reported by
-      HZMI8gkCvPpom0tM)
- * ASTERISK-24719 - ConfBridge recording channels get stuck when
-      recording started/stopped more than once (Reported by Richard
-      Mudgett)
- * ASTERISK-24715 - chan_sip: stale nonce causes failure (Reported
-      by Kevin Harwell)
- * ASTERISK-24728 - tcptls: Bad file descriptor error when
-      reloading chan_sip (Reported by Kevin Harwell)
- * ASTERISK-24676 - Security Vulnerability: URL request injection
-      in libCURL (CVE-2014-8150) (Reported by Matt Jordan)
- * ASTERISK-24711 - DTLS handshake broken with latest OpenSSL
-      versions (Reported by Jared Biel)
- * ASTERISK-24646 - PJSIP changeset 4899 breaks TLS (Reported by
-      Stephan Eisvogel)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c0fe757a54a242ad69da5c601dfcd644ba35557af98ea8fa85c16fb1b38d2c70" name="asterisk" arch="src">
  <version epoch="0" ver="11.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.0-1" date="1428148800">- The Asterisk Development Team has announced the release of Asterisk 11.17.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-17899 - Handle crypto lifetime in SDES-SRTP negotiation
-       (Reported by Dwayne Hubbard)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-24742 - [patch] Fix ast_odbc_find_table function in
-       res_odbc (Reported by ibercom)
-  * ASTERISK-22436 - [patch] No BYE to masqueraded channel on INVITE
-       with replaces (Reported by Eelco Brolman)
-  * ASTERISK-24479 - Enable REF_DEBUG for module references
-       (Reported by Corey Farrell)
-  * ASTERISK-24701 - Stasis: Write timeout on WebSocket fails to
-       fully disconnect underlying socket, leading to events being
-       dropped with no additional information (Reported by Matt Jordan)
-  * ASTERISK-24772 - ODBC error in realtime sippeers when device
-       unregisters under MariaDB (Reported by Richard Miller)
-  * ASTERISK-24451 - chan_iax2: reference leak in sched_delay_remove
-       (Reported by Corey Farrell)
-  * ASTERISK-24799 - [patch] make fails with undefined reference to
-       SSLv3_client_method (Reported by Alexander Traud)
-  * ASTERISK-24787 - [patch] - Microsoft exchange incompatibility
-       for playing back messages stored in IMAP - play_message: No
-       origtime (Reported by Graham Barnett)
-  * ASTERISK-24814 - asterisk/lock.h: Fix syntax errors for non-gcc
-       OSX with 64 bit integers (Reported by Corey Farrell)
-  * ASTERISK-24796 - Codecs and bucket schema's prevent module
-       unload (Reported by Corey Farrell)
-  * ASTERISK-24724 - 'httpstatus' Web Page Produces Incomplete HTML
-       (Reported by Ashley Sanders)
-  * ASTERISK-24797 - bridge_softmix: G.729 codec license held
-       (Reported by Kevin Harwell)
-  * ASTERISK-24800 - Crash in __sip_reliable_xmit due to invalid
-       thread ID being passed to pthread_kill (Reported by JoshE)
-  * ASTERISK-17721 - Incoming SRTP calls that specify a key lifetime
-       fail (Reported by Terry Wilson)
-  * ASTERISK-23214 - chan_sip WARNING message 'We are requesting
-       SRTP for audio, but they responded without it' is ambiguous and
-       wrong in some cases (Reported by Rusty Newton)
-  * ASTERISK-15434 - [patch] When ast_pbx_start failed, both an
-       error response and BYE are sent to the caller (Reported by
-       Makoto Dei)
-  * ASTERISK-18105 - most of asterisk modules are unbuildable in
-       cygwin environment (Reported by feyfre)
-  * ASTERISK-24828 - Fix Frame Leaks (Reported by Kevin Harwell)
-  * ASTERISK-24838 - chan_sip: Locking inversion occurs when
-       building a peer causes a peer poke during request handling
-       (Reported by Richard Mudgett)
-  * ASTERISK-24825 - Caller ID not recognized using
-       Centrex/Distinctive dialing (Reported by Richard Mudgett)
-  * ASTERISK-24739 - [patch] - Out of files -- call fails --
-       numerous files with inodes from under /usr/share/zoneinfo,
-       mostly posixrules (Reported by Ed Hynan)
-  * ASTERISK-23390 - NewExten Event with application AGI shows up
-       before and after AGI runs (Reported by Benjamin Keith Ford)
-  * ASTERISK-24786 - [patch] - Asterisk terminates when playing a
-       voicemail stored in LDAP (Reported by Graham Barnett)
-  * ASTERISK-24808 - res_config_odbc: Improper escaping of
-       backslashes occurs with MySQL (Reported by Javier Acosta)
-  * ASTERISK-20850 - [patch]Nested functions aren't portable.
-       Adapting RAII_VAR to use clang/llvm blocks to get the
-       same/similar functionality. (Reported by Diederik de Groot)
-  * ASTERISK-19470 - Documentation on app_amd is incorrect (Reported
-       by Frank DiGennaro)
-  * ASTERISK-21038 - Bad command completion of "core set debug
-       channel" (Reported by Richard Kenner)
-  * ASTERISK-18708 - func_curl hangs channel under load (Reported by
-       Dave Cabot)
-  * ASTERISK-16779 - Cannot disallow unknown format '' (Reported by
-       Atis Lezdins)
-  * ASTERISK-24876 - Investigate reference leaks from
-       tests/channels/local/local_optimize_away (Reported by Corey
-       Farrell)
-  * ASTERISK-24817 - init_logger_chain: unreachable code block
-       (Reported by Corey Farrell)
-  * ASTERISK-24880 - [patch]Compilation under OpenBSD  (Reported by
-       snuffy)
-  * ASTERISK-24879 - [patch]Compilation fails due to 64bit time
-       under OpenBSD (Reported by snuffy)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24790 - Reduce spurious noise in logs from voicemail -
-       Couldn't find mailbox %s in context (Reported by Graham Barnett)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="03c547390f19fa43cc79e2bdb07e3dc5055426b5641323305edf9135a13e8f42" name="asterisk" arch="src">
  <version epoch="0" ver="11.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.17.1-1" date="1435752000">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 1.8.28, 11.6, and 13.1 and Asterisk 1.8, 11, 12, and 13. The available
- security releases are released as versions 1.8.28.cert-5, 1.8.32.3, 11.6-cert11,
- 11.17.1, 12.8.2, 13.1-cert2, and 13.3.2.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerability:
-
- * AST-2015-003: TLS Certificate Common name NULL byte exploit 
-
-   When Asterisk registers to a SIP TLS device and and verifies the server,
-   Asterisk will accept signed certificates that match a common name other than
-   the one Asterisk is expecting if the signed certificate has a common name
-   containing a null byte after the portion of the common name that Asterisk
-   expected. This potentially allows for a man in the middle attack.
-
- For more information about the details of this vulnerability, please read
- security advisory AST-2015-003, which was released at the same time as this
- announcement.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-1.8.28-cert5
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-1.8.32.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert11
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-12.8.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.1-cert2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.3.2
-
- The security advisory is available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2015-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752001">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="06080713af2bf2ac4ecafeae6097c004635ae877164979912c3ac15ad93f63d9" name="asterisk" arch="src">
  <version epoch="0" ver="11.23.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.18.0-1" date="1435752000">- The Asterisk Development Team has announced the release of Asterisk 11.18.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25112 - Logger: Configuration settings are not reset to
-       default during reload. (Reported by Corey Farrell)
-  * ASTERISK-24887 - [patch]tags in a=crypto lines do not accept 2
-       or more digits (Reported by Makoto Dei)
-  * ASTERISK-24944 - main/audiohook.c change prevents G722 call
-       recording (Reported by Ronald Raikes)
-  * ASTERISK-25083 - Message.c: Message channel becomes saturated
-       with frames leading to spammy log messages (Reported by Jonathan
-       Rose)
-  * ASTERISK-25041 - [patch]Broken column type checking in
-       res_config_mysql addon (Reported by Alexandre Fournier)
-  * ASTERISK-21893 - Segfault after call hangup, in
-       ast_channel_hangupcause_set, at channel_internal_api.c (Reported
-       by Alexandr Gordeev)
-  * ASTERISK-25074 - Regression: Recent clang-related change broke
-       cross compiling of Asterisk (Reported by Sebastian Kemper)
-  * ASTERISK-25042 - asterisk.conf options override command-line
-       options. (Reported by Corey Farrell)
-  * ASTERISK-24442 - Outgoing call files don't work properly when
-       set in the future (Reported by tootai)
-  * ASTERISK-25034 - chan_dahdi: Some telco switches occasionally
-       ignore ISDN RESTART requests. (Reported by Richard Mudgett)
-  * ASTERISK-25038 - Queue log "EXITWITHTIMEOUT" does not always
-       contain waiting time (Reported by Etienne Lessard)
-  * ASTERISK-22708 - res_odbc.conf negative_connection_cache option
-       not respected, failover between DSNs doesn't work (Reported by
-       JoshE)
-  * ASTERISK-25028 - Build System: Unneeded defines in
-       asterisk/buildopts.h (Reported by Corey Farrell)
-  * ASTERISK-19608 - Asterisk-1.8.x  starts rejecting calls with
-       cause code 44 after some time. (Reported by Denis Alberto
-       Martinez)
-  * ASTERISK-24976 - cdr_odbc not include new columns added on 1.8
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25022 - Memory leak setting up DTLS/SRTP calls
-       (Reported by Steve Davies)
-  * ASTERISK-22790 - check_modem_rate() may return incorrect rate
-       for V.27 (Reported by not here)
-  * ASTERISK-23231 - Since 405693 If we have res_fax.conf file set
-       to minrate=2400, then res_fax refuse to load (Reported by David
-       Brillert)
-  * ASTERISK-24955 - res_fax: v.27ter support baud rate of 2400,
-       which is disallowed in res_fax's check_modem_rate (Reported by
-       Matt Jordan)
-  * ASTERISK-24916 - Increasing memory usage when multiple reinvite
-       during call (Reported by Christophe Osuna)
-  * ASTERISK-19538 - Asterisk segfaults on sippeers realtime
-       redundancy (Reported by Alex)
-  * ASTERISK-24749 - ConfBridge: Wrong language on playing
-       conf-hasjoin and conf-hasleft when played to bridge (Reported by
-       Philippe Bolduc)
-  * ASTERISK-24991 - Check for ao2_alloc failure in
-       __ast_channel_internal_alloc (Reported by Corey Farrell)
-  * ASTERISK-24895 - After hangup on the side of the ISDN network no
-       HangupRequest event comes for the dahdi channel. (Reported by
-       Andrew Zherdin)
-  * ASTERISK-24774 - Segfault in ast_context_destroy with
-       extensions.ael and extensions.conf (Reported by Corey Farrell)
-  * ASTERISK-24975 - Enabling 'DEBUG_THREADLOCALS' Causes the Build
-       to Fail (Reported by Ashley Sanders)
-  * ASTERISK-24959 - [patch]CLI command cdr show pgsql status
-       (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-24954 - Git migration: Asterisk version numbers are
-       incompatible with the Test Suite (Reported by Matt Jordan)
-  * ASTERISK-21777 - Asterisk tries to transcode video instead of
-       audio (Reported by Nick Ruggles)
-  * ASTERISK-24380 - core: Native formats are set to h264 with
-       certain audio/video codec configuration, resulting in path
-       translation WARNINGs (Reported by Matt Jordan)
-  * ASTERISK-22352 - [patch] IAX2 custom qualify timer is not taken
-       into account (Reported by Frederic Van Espen)
-  * ASTERISK-24894 - [patch] iax2_poke_noanswer expiration timer too
-       short (Reported by Y Ateya)
-  * ASTERISK-23319 - Segmentation fault in queue_exec at app_queue.c
-       (Reported by Vadim)
-  * ASTERISK-24847 - [security] [patch] tcptls: certificate CN NULL
-       byte prefix bug (Reported by Matt Jordan)
-  * ASTERISK-21211 - chan_iax2 - unprotected access of
-       iaxs[peer-&gt;callno] potentially results in segfault (Reported by
-       Jaco Kroon)
-  * ASTERISK-18032 - [patch] - IPv6 and IPv4 NAT not working
-       (Reported by Christoph Timm)
-  * ASTERISK-24942 - Voicemail API: message is deleted when
-       destination mailbox is at maxmsg (Reported by Scott Griepentrog)
-  * ASTERISK-24932 - Asterisk 13.x does not build with GCC 5.0
-       (Reported by Jeffrey C. Ollie)
-  * ASTERISK-21854 - Long Asterisk-version strings display
-       improperly in the 'Connected to ...' line upon remote console
-       connection (Reported by klaus3000)
-  * ASTERISK-24155 - [patch]Non-portable and non-reliable recursion
-       detection in ast_malloc (Reported by Timo TerÃ¤s)
-  * ASTERISK-24142 - CCSS: crash during shutdown due to device
-       lookup in destroyed container (Reported by David Brillert)
-  * ASTERISK-24683 - Crash in PBX ast_hashtab_lookup_internal during
-       core restart now (Reported by Peter Katzmann)
-  * ASTERISK-24805 - [patch] - ASAN: Race condition
-       (heap-use-after-free) on asterisk closing (Reported by Badalian
-       Vyacheslav)
-  * ASTERISK-24881 - ast_register_atexit should only be used when
-       absolutely needed (Reported by Corey Farrell)
-  * ASTERISK-24864 - app_confbridge: file playback blocks dtmf
-       (Reported by Kevin Harwell)
-  * ASTERISK-14233 - [patch] Buddies are always auto-registered when
-       processing the roster (Reported by Simon Arlott)
-  * ASTERISK-24780 - [patch] - Buddies are always auto-registered
-       when processing the roster (Reported by Simon Arlott)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24744 - Swedish Core Voice prompts (Reported by Tove
-       Hjelm)
-  * ASTERISK-25043 - [patch] Avoiding ERR_remove_state in OpenSSL
-       (Reported by Alexander Traud)
-  * ASTERISK-24917 - [patch] clang compilation warnings (Reported by
-       Diederik de Groot)
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-  * ASTERISK-24965 - cel_pgsql - log_error string references CDR
-       instead of CEL (Reported by Rodrigo Ramirez Norambuena)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
</package>
<package pkgid="b42f0fc0419d5c11851089e148e69587fe5782924b782833e9b172408f514950" name="asterisk" arch="src">
  <version epoch="0" ver="11.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.19.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.19.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25250 - chan_sip - Despite the channel being answered,
-       caller on a call established via Local channel continues to hear
-       ringback (Reported by Etienne Lessard)
-  * ASTERISK-25247 - choppy audio when spying on a g722 channel,
-       chan_sip or chan_pjsip (Reported by hristo)
-  * ASTERISK-24853 - Documentation claims chan_sip outbound
-       registrations support WS or WSS as valid transports (not true)
-       (Reported by PSDK)
-  * ASTERISK-25257 - [patch]channels/sig_pri.h -&gt; sig_pri_span -&gt;
-       force_restart_unavailable_chans in wrong scope (Reported by
-       Patric Marschall)
-  * ASTERISK-25103 - Roundup - investigate Asterisk DTLS crashes
-       (Reported by Rusty Newton)
-  * ASTERISK-22805 - res_rtp_asterisk: Crash when calling
-       BIO_ctrl_pending in dtls_srtp_check_pending when dialed by JSSIP
-       (Reported by Dmitry Burilov)
-  * ASTERISK-24550 - res_rtp_asterisk: Crash in
-       ast_rtp_on_ice_complete during DTLS handshake (Reported by
-       Osaulenko Alexander)
-  * ASTERISK-24651 - [patch] Fix race condition in DTLS (Reported by
-       Badalian Vyacheslav)
-  * ASTERISK-24832 - [patch]DTLS-crashes within openssl  (Reported
-       by Stefan EngstrÃ¶m)
-  * ASTERISK-25127 - DTLS crashes following "Unable to cancel
-       schedule ID" in dtls_srtp_check_pending (Reported by Dade
-       Brandon)
-  * ASTERISK-25213 - [patch]Possibility of deadlock in chan_sip
-       INVITE early Replace code (Reported by Walter Doekes)
-  * ASTERISK-25220 - [patch]Closing of fd -1 in chan_mgcp.c
-       (Reported by Walter Doekes)
-  * ASTERISK-25219 - [patch]Source and destination overlap in memcpy
-       in rtp_engine.c (Reported by Walter Doekes)
-  * ASTERISK-25212 - [patch]Segfault when using DEBUG_FD_LEAKS
-       (Reported by Walter Doekes)
-  * ASTERISK-19277 - [patch]endlessly repeating error: "poll failed:
-       Bad file descriptor" (Reported by Barry Chern)
-  * ASTERISK-25202 - Hints extension state broken between 13.3.2 and
-       13.4 (Reported by cervajs)
-  * ASTERISK-25154 - [patch]fromtag may need to be updated after
-       successful call dialog match (Reported by Damian Ivereigh)
-  * ASTERISK-25139 - Malicious transfer sequence locks up Asterisk
-       (Reported by Gregory Massel)
-  * ASTERISK-25094 - PBX core: Investigate thread safety issues
-       (Reported by Corey Farrell)
-  * ASTERISK-22559 - gcc 4.6 and higher supports weakref attribute
-       but asterisk doesn't detect it. (Reported by ibercom)
-  * ASTERISK-24717 - ASAN: global-buffer-overflow codec_{ilbc | gsm
-       | adpcm | ipc10} (Reported by Badalian Vyacheslav)
-  * ASTERISK-25100 - asterisk coredump if host has an IPv6 address
-       that end with ::80 (Reported by Mark Petersen)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25040 - pbx: Improve performance of reloads by making
-       hint destruction more performant (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947201">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="437665f482c83558cab56d8525af3ceb3197263c5c23e78e704d69577445f0a3" name="asterisk" arch="src">
  <version epoch="0" ver="11.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.20.0-1" date="1445947200">- The Asterisk Development Team has announced the release of Asterisk 11.20.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-  * ASTERISK-25427 - Callerid change does not always emit
-       NewCallerid AMI event (Reported by Ivan Poddubny)
-  * ASTERISK-25407 - Asterisk fails to log to multiple syslog
-       destinations (Reported by Elazar Broad)
-  * ASTERISK-25410 - app_record: RECORDED_FILE variable not being
-       populated (Reported by Kevin Harwell)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25396 - chan_sip: Extremely long callerid name causes
-       invalid SIP (Reported by Walter Doekes)
-  * ASTERISK-25353 - [patch] Transcoding while different in Frame
-       size = Frames lost (Reported by Alexander Traud)
-  * ASTERISK-25227 - No audio at in-band announcements in ooh323
-       channel (Reported by Alexandr Dranchuk)
-  * ASTERISK-25346 - chan_sip: Overwriting answered elsewhere hangup
-       cause on call pickup (Reported by Joshua Colp)
-  * ASTERISK-25215 - Differences in queue.log between Set
-       QUEUE_MEMBER and using PauseQueueMember (Reported by Lorne
-       Gaetz)
-  * ASTERISK-25320 - chan_sip.c: sip_report_security_event searches
-       for wrong or non existent peer on invite (Reported by Kevin
-       Harwell)
-  * ASTERISK-25315 - DAHDI channels send shortened duration DTMF
-       tones. (Reported by Richard Mudgett)
-  * ASTERISK-25312 - res_http_websocket: Terminate connection on
-       fatal cases (Reported by Joshua Colp)
-  * ASTERISK-25265 - [patch]DTLS Failure when calling WebRTC-peer on
-       Firefox 39 - add ECDH support and fallback to prime256v1
-       (Reported by Stefan EngstrÃ¶m)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-25310 - [patch]on FreeBSD also pthread_attr_init()
-       defaults to PTHREAD_EXPLICIT_SCHED (Reported by Guido Falsi)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c76780788da9a89fc98be7811b468ca1e69b7095350d710feac98866b11d69ed" name="asterisk" arch="src">
  <version epoch="0" ver="11.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.0-1" date="1453118400">- The Asterisk Development Team has announced the release of Asterisk 11.21.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25364 - [patch]Issue a TCP connection(kernel) and
-       thread of asterisk is not released (Reported by Hiroaki Komatsu)
-  * ASTERISK-25569 - app_meetme: Audio quality issues (Reported by
-       Corey Farrell)
-  * ASTERISK-25609 - [patch]Asterisk may crash when calling
-       ast_channel_get_t38_state(c) (Reported by Filip Jenicek)
-  * ASTERISK-24146 - [patch]No audio on WebRtc caller side when
-       answer waiting time is more than ~7sec (Reported by Aleksei
-       Kulakov)
-  * ASTERISK-25599 - [patch] SLIN Resampling Codec only 80 msec
-       (Reported by Alexander Traud)
-  * ASTERISK-25616 - Warning with a Codec Module which supports PLC
-       with FEC (Reported by Alexander Traud)
-  * ASTERISK-25610 - Asterisk crash during "sip reload" (Reported by
-       Dudás József)
-  * ASTERISK-25498 - Asterisk crashes when negotiating g729 without
-       that module installed (Reported by Ben Langfeld)
-  * ASTERISK-25476 - chan_sip loses registrations after a while
-       (Reported by Michael Keuter)
-  * ASTERISK-25593 - fastagi: record file closed after sending
-       result (Reported by Kevin Harwell)
-  * ASTERISK-25585 - [patch]rasterisk never hits most of main(), but
-       it's assumed to (Reported by Walter Doekes)
-  * ASTERISK-25552 - hashtab: Improve NULL tolerance (Reported by
-       Joshua Colp)
-  * ASTERISK-25449 - main/sched: Regression introduced by
-       5c713fdf18f causes erroneous duplicate RTCP messages; other
-       potential scheduling issues in chan_sip/chan_skinny (Reported by
-       Matt Jordan)
-  * ASTERISK-25537 - [patch] format-attribute module: RFC or
-       internal defaults? (Reported by Alexander Traud)
-  * ASTERISK-25373 -  add documentation for CALLERID(pres) and also
-       the CONNECTEDLINE and REDIRECTING variants (Reported by Walter
-       Doekes)
-  * ASTERISK-25527 - Quirky xmldoc description wrapping (Reported by
-       Walter Doekes)
-  * ASTERISK-25434 - Compiler flags not reported in 'core show
-       settings' despite usage during compilation (Reported by Rusty
-       Newton)
-  * ASTERISK-25494 - build:  GCC 5.1.x catches some new const, array
-       bounds and missing paren issues (Reported by George Joseph)
-  * ASTERISK-7803 - [patch] Update the maximum packetization values
-       in frame.c (Reported by dea)
-  * ASTERISK-25461 - Nested dialplan #includes don't work as
-       expected. (Reported by Richard Mudgett)
-  * ASTERISK-25455 - Deadlock of PJSIP realtime over
-       res_config_pgsql  (Reported by mdu113)
-  * ASTERISK-25135 - [patch]RTP Timeout hangup cause code missing
-       (Reported by Olle Johansson)
-  * ASTERISK-25400 - Hints broken when "CustomPresence" doesn't
-       exist in AstDB (Reported by Andrew Nagy)
-  * ASTERISK-25443 - [patch]IPv6 - Potential issue in via header
-       parsing (Reported by ffs)
-  * ASTERISK-25391 - AMI GetConfigJSON returns invalid JSON
-       (Reported by Bojan Nemčić)
-  * ASTERISK-25438 - res_rtp_asterisk: ICE role message even when
-       ICE is not enabled (Reported by Joshua Colp)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24718 - [patch]Add inital support of "sanitize" to
-       configure (Reported by Badalian Vyacheslav)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5f9825388c7d8c637cd9ab150a0dd6483ea817205b085ebea2fc43d2d2b47fff" name="asterisk" arch="src">
  <version epoch="0" ver="11.25.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-1" date="1454673600">- The Asterisk Development Team has announced security releases for Certified
- Asterisk 11.6 and 13.1 and Asterisk 11 and 13. The available security releases
- are released as versions 11.6-cert12, 11.21.1, 13.1-cert3, and 13.7.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2016-001: BEAST vulnerability in HTTP server
-
-   The Asterisk HTTP server currently has a default configuration which allows
-   the BEAST vulnerability to be exploited if the TLS functionality is enabled.
-   This can allow a man-in-the-middle attack to decrypt data passing through it.
-
- * AST-2016-002: File descriptor exhaustion in chan_sip
-
-   Setting the sip.conf timert1 value to a value higher than 1245 can cause an
-   integer overflow and result in large retransmit timeout times. These large
-   timeout values hold system file descriptors hostage and can cause the system
-   to run out of file descriptors.
-
- * AST-2016-003: Remote crash vulnerability receiving UDPTL FAX data.
-
-   If no UDPTL packets are lost there is no problem. However, a lost packet
-   causes Asterisk to use the available error correcting redundancy packets. If
-   those redundancy packets have zero length then Asterisk uses an uninitialized
-   buffer pointer and length value which can cause invalid memory accesses later
-   when the packet is copied.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.21.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.1-cert3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.7.1
-
- The security advisories are available at:
-
-  * http://downloads.asterisk.org/pub/security/AST-2016-001.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-002.pdf
-  * http://downloads.asterisk.org/pub/security/AST-2016-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="765fae1eed3f8ead5cfd05c3868f5f798122f4da7adfd36735556c0052bbbdc4" name="asterisk" arch="src">
  <version epoch="0" ver="11.25.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.1-2" date="1455105600">- Added tmpfiles.d systemd configuration file to create /var/run/asterisk with
- correct permissions at boot time.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
</package>
<package pkgid="a4f014d37831a5f506a4dcdbcd9e20952707a5672e647ba9bfb93135be1773c7" name="asterisk" arch="src">
  <version epoch="0" ver="11.25.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.21.2-1" date="1456401600">- The Asterisk Development Team has announced the release of Asterisk 11.21.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.21.2 resolves an issue reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25770 - Check for OpenSSL defines before trying to use
-       them. (Reported by Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.21.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
</package>
<package pkgid="70b20fc34673ca5592a6324360fd540baa6a1d41c307f6ea2af56d35927a18c7" name="asterisk" arch="src">
  <version epoch="0" ver="11.25.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.22.0-1" date="1459339200">- The Asterisk Development Team has announced the release of Asterisk 11.22.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-25857 - func_aes: incorrect use of strlen() leads to
-       data corruption (Reported by Gianluca Merlo)
-  * ASTERISK-25321 - [patch]DeadLock ChanSpy with call over Local
-       channel (Reported by Filip Frank)
-  * ASTERISK-25800 - [patch] Calculate talktime when is first call
-       answered (Reported by Rodrigo Ramirez Norambuena)
-  * ASTERISK-25272 - [patch]The ICONV dialplan function sometimes
-       returns garbage (Reported by Etienne Lessard)
-  * ASTERISK-20987 - non-admin users, who join muted conference are
-       not being muted (Reported by hristo)
-  * ASTERISK-24972 - Transport Layer Security (TLS) Protocol BEAST
-       Vulnerability - Investigate vulnerability of HTTP server
-       (Reported by Alex A. Welzl)
-  * ASTERISK-25603 - [patch]udptl: Uninitialized lengths and bufs in
-       udptl_rx_packet cause ast_frdup crash (Reported by Walter
-       Doekes)
-  * ASTERISK-25742 - Secondary IFP Packets can result in accessing
-       uninitialized pointers and a crash (Reported by Torrey Searle)
-  * ASTERISK-25397 - [patch]chan_sip: File descriptor leak with
-       non-default timert1 (Reported by Alexander Traud)
-  * ASTERISK-25730 - build:  make uninstall after make distclean
-       tries to remove root (Reported by George Joseph)
-  * ASTERISK-25722 - ASAN &amp; testsute: stack-buffer-overflow in
-       sip_sipredirect (Reported by Badalian Vyacheslav)
-  * ASTERISK-25714 - ASAN:heap-buffer-overflow in logger.c (Reported
-       by Badalian Vyacheslav)
-  * ASTERISK-24801 - ASAN: ast_el_read_char stack-buffer-overflow
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25701 - core: Endless loop in "core show
-       taskprocessors" (Reported by ibercom)
-  * ASTERISK-25700 - main/config: Clean config maps on shutdown.
-       (Reported by Corey Farrell)
-  * ASTERISK-25690 - Hanging up when executing connected line sub
-       does not cause hangup (Reported by Joshua Colp)
-  * ASTERISK-25687 - res_musiconhold: Concurrent invocations of 'moh
-       reload' cause a crash (Reported by Sean Bright)
-  * ASTERISK-25394 - pbx: Incorrect device and presence state when
-       changing hint details (Reported by Joshua Colp)
-  * ASTERISK-25640 - pbx: Deadlock on features reload and state
-       change hint. (Reported by Krzysztof Trempala)
-  * ASTERISK-25681 - devicestate: Engine thread is not shut down
-       (Reported by Corey Farrell)
-  * ASTERISK-25680 - manager: manager_channelvars is not cleaned at
-       shutdown (Reported by Corey Farrell)
-  * ASTERISK-25679 - res_calendar leaks scheduler. (Reported by
-       Corey Farrell)
-  * ASTERISK-25677 - pbx_dundi: leaks during failed load. (Reported
-       by Corey Farrell)
-  * ASTERISK-25673 - res_crypto leaks CLI entries (Reported by Corey
-       Farrell)
-  * ASTERISK-25647 - bug of cel_radius.c: wrong point of
-       ADD_VENDOR_CODE (Reported by Aaron An)
-  * ASTERISK-25614 - DTLS negotiation delays (Reported by Dade
-       Brandon)
-  * ASTERISK-25442 - using realtime (mysql) queue members are never
-       updated in wait_our_turn function (app_queue.c)  (Reported by
-       Carlos Oliva)
-  * ASTERISK-25624 - AMI Event OriginateResponse bug (Reported by
-       sungtae kim)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24813 - asterisk.c: #if statement in listener()
-       confuses code folding editors (Reported by Corey Farrell)
-  * ASTERISK-25767 - [patch] Add check to configure for sanitizes
-       (Reported by Badalian Vyacheslav)
-  * ASTERISK-25068 - Move commonly used FreePBX extra sounds to the
-       core set (Reported by Rusty Newton)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f613135e052bfc6c4dc035a1aca08e6b15dbafb53c1366a9c4db0deecea4cbd1" name="asterisk" arch="src">
  <version epoch="0" ver="11.25.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.0-1" date="1469102400">- The Asterisk Development Team has announced the release of Asterisk 11.23.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.23.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26141 - res_fax: fax_v21_session_new leaks reference to v21_details
-      (Reported by Corey Farrell)
- * ASTERISK-26140 - res_rtp_asterisk: gcc 6 caught a self-comparison (Reported
-      by George Joseph)
- * ASTERISK-26138 - chan_unistim:  Under FreeBSD, chan_unistim generates a
-      compile error (Reported by George Joseph)
- * ASTERISK-26130 - [patch] WebRTC: Should use latest DTLS version. (Reported by
-      Alexander Traud)
- * ASTERISK-26126 - [patch] leverage 'bindaddr' for TLS in http.conf (Reported
-      by Alexander Traud)
- * ASTERISK-26069 - Asterisk truncates To: header, dropping the closing '&gt;'
-      (Reported by Vasil Kolev)
- * ASTERISK-26097 - [patch] CLI: show maximum file descriptors (Reported by
-      Alexander Traud)
- * ASTERISK-24436 - Missing header in res/res_srtp.c when compiling against
-      libsrtp-1.5.0 (Reported by Patrick Laimbock)
- * ASTERISK-26091 - [patch] ar cru creates warning, instead use ar cr (Reported
-      by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26034 - T.38 passthrough problem behind firewall due to early
-      nosignal packet (Reported by George Joseph)
- * ASTERISK-26030 - call cut because of double Session-Expires header in
-      re-invite after proxy authentication is required (Reported by George
-      Joseph)
- * ASTERISK-26008 - app_followme does not delete recorded name prompt (Reported
-      by Tzafrir Cohen)
- * ASTERISK-24463 - Voicemail email address corrupt or not sent when message is
-      in the process of being recorded during reload (Reported by John Campbell)
- * ASTERISK-25917 - [patch]app_voicemail: passwordlocation=spooldir only works
-      if you manually add secret.conf yourself (Reported by Jonathan R. Rose)
- * ASTERISK-25954 - Manager QueueSummary and QueueStatus Actions are case
-      sensitive to QueueName (Reported by Javier Acosta)
- * ASTERISK-16115 - [patch] problem with ringinuse=no, queue members receive
-      sometimes two calls (Reported by nik600)
- * ASTERISK-25934 - chan_sip should not require sipregs or updateable sippeers
-      table unless rt (Reported by Jaco Kroon)
- * ASTERISK-25888 - Frequent segfaults in function can_ring_entry() of
-      app_queue.c (Reported by Sébastien Couture)
- * ASTERISK-25874 - app_voicemail: Stack buffer overflow in
-      test_voicemail_notify_endl (Reported by Badalian Vyacheslav)
- * ASTERISK-25912 - chan_local passes AST_CONTROL_PVT_CAUSE_CODE without adding
-      them to the local hangupcauses via ast_channel_hangupcause_hash_set
-      (Reported by Jaco Kroon)
- * ASTERISK-25407 - Asterisk fails to log to multiple syslog destinations
-      (Reported by Elazar Broad)
- * ASTERISK-25510 - [patch]Log to syslog failing (Reported by Michael Newton)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-25444 - [patch]Music On Hold Warning misleading (Reported by Conrad
-      de Wet)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.23.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.23.1-1" date="1473336000">- Security release 11.23.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.0-1" date="1477396800">- The Asterisk Development Team has announced the release of Asterisk 11.24.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26438 - [patch] chan_sip: auto_force_rport: No NAT = No Symmetric
-      Response. (Reported by Alexander Traud)
- * ASTERISK-18232 - Broken REGISTER sent to IPv4 server when bindaddr=[::]
-      (Reported by Jacek)
- * ASTERISK-26359 - [patch] cdr_mysql: fails to use UTC if so instructed
-      (Reported by Tzafrir Cohen)
- * ASTERISK-19968 - TCP Session-Timers not dropping call (Reported by Aaron
-      Hamstra)
- * ASTERISK-26360 - app_queue: "queue show" output gets "failed to extend from
-      240 to 327" msgs. (Reported by Richard Mudgett)
- * ASTERISK-26272 - chan_sip: File descriptors leak (UDP sockets) (Reported by
-      Etienne Lessard)
- * ASTERISK-26288 - followme: fails to reset config items to default values on
-      reload (Reported by Tzafrir Cohen)
- * ASTERISK-26282 - AEL: macro-call in Dial application, macro "lacks 's'
-      extension" (Reported by chris de rock)
- * ASTERISK-26226 - pbx: Asterisk crash on AMI action "ShowDialplan" when
-      there's a circular dependency between contexts (Reported by Etienne
-      Lessard)
- * ASTERISK-26299 - app_queue: Queue application sometimes stops calling members
-      with Local interface (Reported by Etienne Lessard)
- * ASTERISK-26306 - channel: Hang-up crashes, chan_pjsip not cleaning up
-      properly (Reported by Alexander Traud)
- * ASTERISK-26203 - res_fax: Deadlock when using FAXOPT(gateway)=yes with Local
-      channels (Reported by Etienne Lessard)
- * ASTERISK-24822 - Deadlock: Fax Gateway framehook creates locking inversion in
-      T.38 query option with features bridging code (Reported by David Brillert)
- * ASTERISK-22732 - Deadlock potential in res_fax and CCSS with local channels.
-      (Reported by Richard Mudgett)
- * ASTERISK-24841 - ConfBridge: Strange sampling rates chosen when channels have
-      multiple native formats (Reported by Matt Jordan)
- * ASTERISK-24425 - [patch] jabber/xmpp to use TLS instead of SSLv3, security
-      fix POODLE (CVE-2014-3566) (Reported by abelbeck)
- * ASTERISK-25706 - pbx: Abort asterisk on features reload (handle_hint_change)
-      (Reported by Krzysztof Trempala)
- * ASTERISK-26233 - pbx: Failure to remove inconsistent extension names
-      (Reported by Corey Farrell)
- * ASTERISK-26267 - ast_register_atexit callbacks should be run on failed
-      startup. (Reported by Corey Farrell)
- * ASTERISK-26265 - Errors ignored from some parts of system initialization.
-      (Reported by Corey Farrell)
- * ASTERISK-25996 - Remove "live_dangerously" requirement on DB(read) (Reported
-      by Andrew Nagy)
- * ASTERISK-26237 - Fax is detected on regular calls. (Reported by Richard
-      Mudgett)
- * ASTERISK-23013 - [patch] Deadlock between 'sip show channels' command and
-      attended transfer handling (Reported by Ben Smithurst)
- * ASTERISK-26211 - Unit tests: AST_TEST_DEFINE should be used in conditional
-      code. (Reported by Corey Farrell)
- * ASTERISK-26207 - [patch] sRTP: Count a roll-over of the sequence number even
-      on lost packets. (Reported by Alexander Traud)
- * ASTERISK-26038 - 'make install' doesn't seem to install OS/X init files
-      (Reported by Tzafrir Cohen)
- * ASTERISK-26133 - app_queue: Queue members receive multiple calls (Reported by
-      Richard Miller)
- * ASTERISK-26196 - pbx: Time based includes can leak timezone string (Reported
-      by Corey Farrell)
- * ASTERISK-25659 - res_rtp_asterisk: ECDH not negotiated causing DTLS failure
-      occurred on RTP instance (Reported by Edwin Vandamme)
- * ASTERISK-26046 - [patch] Avoid obsolete warnings on autoconf. (Reported by
-      Alexander Traud)
- * ASTERISK-25289 - Build System does not respect CFLAGS and CXXFLAGS when
-      building menuselect (Reported by Jeffrey Walton)
- * ASTERISK-26119 - [patch] fix: memory leaks, resource leaks, out of bounds and
-      bugs (Reported by Alexei Gradinari)
- * ASTERISK-26179 - chan_sip: Second T.38 request fails (Reported by Joshua
-      Colp)
- * ASTERISK-26157 - Build: Fix errors highlighted by GCC 6.x (Reported by George
-      Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- * ASTERISK-26220 - Add support for noreturn function attributes. (Reported by
-      Corey Farrell)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.24.1-1" date="1477569600">- The Asterisk Development Team has announced the release of Asterisk 11.24.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.24.1 resolves an issue reported by the community and
- would have not been possible without your participation. Thank you!
-
- The following is the issue resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.24.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.0-1" date="1480507200">- The Asterisk Development Team has announced the release of Asterisk 11.25.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 11.25.0 resolves several issues reported by the
- community and would have not been possible without your participation. Thank
- you!
-
- The following are the issues resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- * ASTERISK-26503 - app_voicemail: Asterisk crashes when MailboxExists is used
-      (Reported by Doug Lytle)
- * ASTERISK-26480 - [patch] CLI: core set debug: Auto-completes File not Module
-      (Reported by Alexander Traud)
- * ASTERISK-26356 - menuselect: invalid test for GTK2 (Reported by Tzafrir
-      Cohen)
- * ASTERISK-26462 - [patch] app_queue: While using queues with realtime, setting
-      back to an empty context doesn't stop the exit key usage (Reported by
-      Leandro Dardini)
- * ASTERISK-26457 - [patch] force_rport,auto_comedia: No NAT detection
-      triggered. (Reported by Alexander Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-11.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-1" date="1486728000">- The Asterisk Development Team has announced security releases for Asterisk
- 11, 13, 14, and Certified Asterisk 11.6 and 13.8. The available security
- releases are released as versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16,
- and 13.8-cert4.
-
- These releases are available for immediate download at:
-
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of versions 13.13.1 and 14.2.1 resolve the following security
- vulnerability:
-
- * AST-2016-008: Crash on SDP offer or answer from endpoint using Opus
-
- If an SDP offer or answer is received with the Opus codec and with the format
- parameters separated using a space the code responsible for parsing will
- recursively call itself until it crashes. This occurs as the code does not
- properly handle spaces separating the parameters.
-
- This does NOT require the endpoint to have Opus configured in Asterisk. This
- also does not require the endpoint to be authenticated. If guest is enabled
- for chan_sip or anonymous in chan_pjsip an SDP offer or answer is still
- processed and the crash occurs.
-
- The release of versions 11.25.1, 13.13.1, 14.2.1, 11.6-cert16 and 13.8-cert4
- resolve the following security vulnerability:
-
- * AST-2016-009: Remote unauthenticated sessions in chan_sip
-
- The chan_sip channel driver has a liberal definition for whitespace when
- attempting to strip the content between a SIP header name and a colon
- character. Rather than following RFC 3261 and stripping only spaces and
- horizontal tabs, Asterisk treats any non-printable ASCII character as if it
- were whitespace. This means that headers such as
-
- Contact\x01:
-
- will be seen as a valid Contact header.
-
- This mostly does not pose a problem until Asterisk is placed in tandem with
- an authenticating SIP proxy. In such a case, a crafty combination of valid
- and invalid To headers can cause a proxy to allow an INVITE request into
- Asterisk without authentication since it believes the request is an in-dialog
- request. However, because of the bug described above, the request will look
- like an out-of-dialog request to Asterisk. Asterisk will then process the
- request as a new call. The result is that Asterisk can process calls from
- unvetted sources without any authentication.
-
- If you do not use a proxy for authentication, then this issue does not affect
- you. If your proxy is dialog-aware (meaning that the proxy keeps track of what
- dialogs are currently valid), then this issue does not affect you. If you use
- chan_pjsip instead of chan_sip, then this issue does not affect you.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-11.25.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-13.13.1
- http://downloads.asterisk.org/pub/telephony/asterisk/release s/ChangeLog-14.2.1
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-11.6-cert16
- http://downloads.asterisk.org/pub/telephony/certified-asteri sk/releases/ChangeLog-certified-13.8-cert4
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2016-008.pdf
- * http://downloads.asterisk.org/pub/security/AST-2016-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-2" date="1494936000">- Added OPUS codec support
- Added VP8 passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.1-3" date="1505304000">- Added RTCP security bugfix. More info at https://github.com/EnableSecurity/advisories/tree/master/ES2017-04-asterisk-rtp-bleed</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.2-1" date="1505304001">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.2, 13.17.1, 14.6.1, 11.6-cert17, and 13.13-cert5.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-005 (applied to all released versions): The “strictrtp” option in
-   rtp.conf enables a feature of the RTP stack that learns the source address of
-   media for a session and drops any packets that do not originate from the
-   expected address. This option is enabled by default in Asterisk 11 and above.
-
-   The "nat" and "rtp_symmetric" options for chan_sip and chan_pjsip
-   respectively enable symmetric RTP support in the RTP stack. This uses the
-   source address of incoming media as the target address of any sent media.
-   This option is not enabled by default but is commonly enabled to handle
-   devices behind NAT.
-
-   A change was made to the strict RTP support in the RTP stack to better
-   tolerate late media when a reinvite occurs. When combined with the symmetric
-   RTP support this introduced an avenue where media could be hijacked. Instead
-   of only learning a new address when expected the new code allowed a new
-   source address to be learned at all times.
-
-   If a flood of RTP traffic was received the strict RTPsupport would allow the
-   new address to provide media and with symmetric RTP enabled outgoing traffic
-   would be sent to this new address, allowing the media to be hijacked.
-   Provided the attacker continued to send traffic they would continue to
-   receive traffic as well.
-
- * AST-2017-006 (applied to all released versions): The app_minivm module has an
-   "externnotify" program configuration option that is executed by the
-   MinivmNotify dialplan application. The application uses the caller-id name
-   and number as part of a built string passed to the OS shell for
-   interpretation and execution. Since the caller-id name and number can come
-   from an untrusted source, a crafted caller-id name or number allows an
-   arbitrary shell command injection.
-
- * AST-2017-007 (applied only to 13.17.1 and 14.6.1): A carefully crafted URI in
-   a From, To or Contact header could cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.1
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-11.6-cert17
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-13.13-cert5
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-005.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-006.pdf
- * http://downloads.asterisk.org/pub/security/AST-2017-007.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 11.25.3-1" date="1505822400">- The Asterisk Development Team has announced security releases for Asterisk 11,
- 13, and 14, and for Certified Asterisk 11.6 and 13.13. The available security
- release versions are 11.25.3, 13.17.2, 14.6.2, 11.6-cert18, and 13.13-cert6.
-
- These releases are available for immediate download at
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/
-
- The release of these versions resolves the following security vulnerabilities:
-
- * AST-2017-008: Insufficient RTCP packet validation could allow reading stale
-   buffer contents and when combined with the “nat” and “symmetric_rtp” options
-   allow redirecting where Asterisk sends the next RTCP report.
-
- The RTP stream qualification to learn the source address of media always
- accepted the first RTP packet as the new source and allowed what AST-2017-005
- was mitigating. The intent was to qualify a series of packets before accepting
- the new source address.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-11.25.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.17.2
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.6.2
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-11.6-cert18
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.13-cert6
-
- The security advisories are available at:
-
- * http://downloads.asterisk.org/pub/security/AST-2017-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7f78e991f29368fe83dceb9a2215bdc8bc0949914f632953d836f0e08c91ab40" name="asterisk" arch="src">
  <version epoch="0" ver="15.0.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 14.5.0-4" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-1" date="1504180800">- Update to upstream 14.6.0 release
- Re-enable radius sub-packages</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-2" date="1504267200">- Add perl to BuildRequires</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267201">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267202">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267203">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="16712e03f2a09db341af4c59a29db0100e6c6e01fa24e0ae8adaa2b345ca2ef0" name="asterisk" arch="src">
  <version epoch="0" ver="15.1.0" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-1" date="1504180800">- Update to upstream 14.6.0 release
- Re-enable radius sub-packages</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-2" date="1504267200">- Add perl to BuildRequires</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267201">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267202">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267203">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f7e0cebeab418a8a8874fb212a51bdb703e1200bcea4d97e7dd6ddf7d761c6cd" name="asterisk" arch="src">
  <version epoch="0" ver="15.1.1" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.0-2" date="1504267200">- Add perl to BuildRequires</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267201">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267202">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267203">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9a4bba1cca1f259abf910ab227faffa1622ad9086d22b035f9fb9cb7e9b2cee3" name="asterisk" arch="src">
  <version epoch="0" ver="15.1.2" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-1" date="1504267200">- Update to upstream 14.6.1 release
- Solves AST-2017-005, AST-2017-006, and AST-2017-007 security issues</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267201">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267202">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c9fadf8eedfb78dbe0a9305fb83aad7af968f73205113198086a3acf241fa7fe" name="asterisk" arch="src">
  <version epoch="0" ver="15.1.3" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-3" date="1504267200">- Fix MySQL header path (due to change in mariadb-devel patckage)</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267201">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6e8a328e6d824979d7d72493ed42648dfa75f8cea181dcd8a467dc62f1652487" name="asterisk" arch="src">
  <version epoch="0" ver="15.1.4" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-4" date="1504267200">- Disable corosync modules until corosync works in ppc64le again</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a7f760c64f2e11a21112d516dfc6d6724e83d72940dd70099791f4a8560e5367" name="asterisk" arch="src">
  <version epoch="0" ver="15.1.5" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-5" date="1504440000">- Add dependency on unbound-devel for res_resolver_unbound</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6781cef00e8ef066dad265f4e947d8d822f634f8a98d5a70a4bae464454795e4" name="asterisk" arch="src">
  <version epoch="0" ver="15.2.0" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.1-6" date="1505304000">- Re-enable corosync, see RHBZ#1478089</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b5e20bc2efaab7a9ab303389f2e3ee253e28ad14343abab241fbb4cb4af691e9" name="asterisk" arch="src">
  <version epoch="0" ver="15.2.1" rel="1.el7_9"/>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 14.6.2-1" date="1505995200">- Update to upstream 14.6.2 release</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="42120b9ee1b7053fab94b29087a3852cd51a13ad8a580f111da642c81afd03cf" name="asterisk" arch="src">
  <version epoch="0" ver="15.2.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.0.0-1" date="1507204800">- SPEC originally by Jared Smith &lt;jsmith@fedoraproject.org&gt; (modified MariaDB requirement.)
-
- The Asterisk Development Team would like to announce the release of Asterisk 15.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-26230] -
- 		[patch] res_pjsip_mwi: unsolicited mwi could block PJSIP taskprocessor on startup
- (Reported by Alexei Gradinari)
- [ASTERISK-27043] -
- 		Core/BuildSystem: Add defines to fix build with LibreSSL
- (Reported by Guido Falsi)
- [ASTERISK-27042] -
- 		Unpatched asterisk sources fail to build on FreeBSD due to missing crypt.h file
- (Reported by Guido Falsi)
- [ASTERISK-26419] -
- 		audiohooks: Remove redundant codec translations when using audiohooks
- (Reported by Michael Walton)
- [ASTERISK-26976] -
- 		libsrtp-2.x.x support
- (Reported by Alex)
- [ASTERISK-27014] -
- 		configurable busy_timeout in sqlite backends
- (Reported by Marek Cervenka)
- [ASTERISK-26124] -
- 		res_agi: Set audio format for EAGI audio stream
- (Reported by John Fawcett)
- [ASTERISK-26088] -
- 		Investigate heavy memory utilization by res_pjsip_pubsub
- (Reported by Richard Mudgett)
- [ASTERISK-26427] -
- 		res_hep_rtcp: Asterisk Master will report channel name with res_hep_rtcp when using chan_sip
- (Reported by Nir Simionovich (GreenfieldTech - Israel))
- [ASTERISK-26932] -
- 		[patch] SIP/SDP: No rtpmap for static RTP payload IDs
- (Reported by Alexander Traud)
- [ASTERISK-26864] -
- 		res_pjsip_session: Add support for overlap dialling
- (Reported by Richard Begg)
- [ASTERISK-26846] -
- 		chan_sip: Add rtcp-mux support
- (Reported by Sean Bright)
- [ASTERISK-26568] -
- 		pbx_spool: OUTGOING_RETRY variable
- (Reported by Roman Shubovich)
- [ASTERISK-26292] -
- 		app_confbridge: 3D-Conferencing via Binaural Synthesis
- (Reported by Dennis Guse)
- [ASTERISK-23828] -
- 		pjsip - Need a command to list active SIP subscriptions
- (Reported by Rusty Newton)
- [ASTERISK-26559] -
- 		app_queue: New service level calculation
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26658] -
- 		Add ability for dialplan show to display filenames/line numbers of registered extensions
- (Reported by Jonathan R. Rose)
- [ASTERISK-26527] -
- 		Testsuite: increase timeout to check "core fullybooted wait" up to 30 sec
- (Reported by Badalian Vyacheslav)
- [ASTERISK-22992] -
- 		[patch]Asterisk app_originate doesn't allow setting Caller*ID on the originating channel
- (Reported by Anthony Messina)
- [ASTERISK-26624] -
- 		res_calendar_caldav: Add support for gmail
- (Reported by Eduardo Scudeller Libardi)
- [ASTERISK-26562] -
- 		app_controlplayback: Transmit Silence on ControlPlayback pause
- (Reported by Mikheili Dautashvili)
- [ASTERISK-24517] -
- 		TLS support for Solaris, Ming and non-glibc Linux systems
- (Reported by Timo Teräs)
- [ASTERISK-26540] -
- 		cdr_radius: use radcli instead of freeradius-client
- (Reported by Tzafrir Cohen)
- [ASTERISK-26558] -
- 		app_queue: add variable to know if the call is not answered after a queue
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26176] -
- 		chan_sip: Add AccountCode to AMI PeerEntry
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26217] -
- 		[patch] Codec 2 Mode 2400
- (Reported by Alexander Traud)
- [ASTERISK-26538] -
- 		codec_opus: Add sample to configs/samples/codecs.conf.sample
- (Reported by Kevin Harwell)
- [ASTERISK-26488] -
- 		ARI: Add 'ari show app', 'ari show apps', and 'ari set debug' CLI commands
- (Reported by Matt Jordan)
- [ASTERISK-26418] -
- 		res_rtp_asterisk: Speed up ICE resolution by blacklisting host subnets that are not involved in RTP
- (Reported by Michael Walton)
- [ASTERISK-26422] -
- 		[patch] Force calendars to do new fetch after module reload
- (Reported by Ludovic Gasc (Eyepea))
- [ASTERISK-26398] -
- 		core: Remove ABI differences of LOW_MEMORY
- (Reported by Corey Farrell)
- [ASTERISK-26409] -
- 		codec_opus: Update Asterisk to support the translation codec.
- (Reported by Kevin Harwell)
- [ASTERISK-26289] -
- 		Announcer channels in ConfBridges cause inefficiencies
- (Reported by Mark Michelson)
- [ASTERISK-26321] -
- 		ARI : Add reason answered_elsewhere to channel hangup
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-25980] -
- 		[patch]res_fax: set FAXMODE variable to let dialplan know what fax transport was used
- (Reported by Alexei Gradinari)
- [ASTERISK-26218] -
- 		[patch] iLBC 20
- (Reported by Alexander Traud)
- [ASTERISK-26190] -
- 		[patch] SRTP: Enable AES-256 and AES-GCM.
- (Reported by Alexander Traud)
- [ASTERISK-26220] -
- 		Add support for noreturn function attributes.
- (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27143] -
- 		bridge_softmix / res_rtp_asterisk: Fix packet loss and renegotiation issues.
- (Reported by Joshua Colp)
- [ASTERISK-27142] -
- 		sounds: Conflict between files in asterisk-sounds-core-1.6 and asterisk-sounds-extra-1.5
- (Reported by Corey Farrell)
- [ASTERISK-25810] -
- 		say.c calls for sounds in the subdir "digits" that don't exist (in Core). SayUnixTime or other Say... apps will fail out when they call these sounds.
- (Reported by Nicolas Riendeau)
- [ASTERISK-27136] -
- 		bridge_softmix: Don't reorder SFU streams
- (Reported by Joshua Colp)
- [ASTERISK-27134] -
- 		bridge_softmix: Reuse any removed streams for video
- (Reported by Joshua Colp)
- [ASTERISK-27133] -
- 		res_rtp_asterisk: RTCP does not use ICE when RTCP-MUX in use
- (Reported by Joshua Colp)
- [ASTERISK-27123] -
- 		confbridge: Name recordings are left on filesystem
- (Reported by Sergej Kasumovic)
- [ASTERISK-27122] -
- 		chan_iax2: On reload MWI taskprocessors keep adding up
- (Reported by Sergej Kasumovic)
- [ASTERISK-26807] -
- 		sounds: New 3-D Binaural audio features require new sound prompts
- (Reported by Rusty Newton)
- [ASTERISK-25816] -
- 		French conf-adminmenu, conf-usermenu prompts differ in content from the English files
- (Reported by Benoit Duverger)
- [ASTERISK-26274] -
- 		Resolve open sounds issues and then create a new sounds release (1.5.1? or 1.6?)
- (Reported by Rusty Newton)
- [ASTERISK-27118] -
- 		res_pjsip_session / res_rtp_asterisk: Add support for BUNDLE
- (Reported by Joshua Colp)
- [ASTERISK-27036] -
- 		res_pjsip: Asterisk crashes when an extension tries to use PJSIP trunk with from_user containing '@'
- (Reported by Maxim Vasilev)
- [ASTERISK-27023] -
- 		res_rtp_asterisk: Deadlock when TURN session in use
- (Reported by Jatin Jain)
- [ASTERISK-27106] -
- 		[patch] autodomain (SIP Domain Support): Add only really different domain with TLS.
- (Reported by Alexander Traud)
- [ASTERISK-27093] -
- 		ODBC deadlocks when app_directory tries to play back non-existent voicemail greeting
- (Reported by James Terhune)
- [ASTERISK-27100] -
- 		channel: ast_waitfordigit_full fails to clear flag in an error branch.
- (Reported by Corey Farrell)
- [ASTERISK-27090] -
- 		PJSIP: Deadlock using TCP transport
- (Reported by Richard Mudgett)
- [ASTERISK-26997] -
- 		Create an StreamEcho dialplan application
- (Reported by Kevin Harwell)
- [ASTERISK-27076] -
- 		chan_pjsip: Add support for multiple streams
- (Reported by Joshua Colp)
- [ASTERISK-27088] -
- 		res_rtp_asterisk: Better handle ICE renegotiation and unidirectional negotiation
- (Reported by Joshua Colp)
- [ASTERISK-26978] -
- 		rtp: Crash in ast_rtp_codecs_payload_code()
- (Reported by Ross Beer)
- [ASTERISK-25665] -
- 		Duplicate logging in queue log for EXITEMPTY events
- (Reported by Ove Aursand)
- [ASTERISK-27065] -
- 		call hangup after leaving app_queue
- (Reported by Marek Cervenka)
- [ASTERISK-24052] -
- 		app_voicemail reloads result in leaked IMAP sockets.
- (Reported by Louis Jocelyn Paquet)
- [ASTERISK-27074] -
- 		core_local: local channel data not being properly unref'ed and unlocked
- (Reported by Kevin Harwell)
- [ASTERISK-27075] -
- 		bridge: stuck channel(s) after failed attended transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27060] -
- 		Comment typo format_g729.c
- (Reported by Matthew Fredrickson)
- [ASTERISK-27041] -
- 		Core/PBX: [patch] Deadlock between dialplan execution and application unregistration
- (Reported by Frederic LE FOLL)
- [ASTERISK-25370] -
- 		res_corosync segfaults at startup with corosync version &gt; 2.x
- (Reported by mdu113)
- [ASTERISK-27026] -
- 		res_ari: Crash when no ari.conf configuration file exists
- (Reported by Ronald Raikes)
- [ASTERISK-27016] -
- 		Crash occurs when a channel in a 'mixing,dtmf_events' bridge is muted multiple times.
- (Reported by Chris Howard)
- [ASTERISK-27057] -
- 		Seg Fault in ast_sorcery_object_get_id at sorcery.c
- (Reported by Ryan Smith)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-27022] -
- 		res_rtp_asterisk: Incorrect SSRC change for RTCP component
- (Reported by Michael Walton)
- [ASTERISK-26923] -
- 		bridging: T.38 request is lost when channels are added to bridge
- (Reported by Torrey Searle)
- [ASTERISK-27053] -
- 		res_pjsip_refer/session: Calls dropped during transfer
- (Reported by Kevin Harwell)
- [ASTERISK-27052] -
- 		Asterisk build process fails with flag --with-pjproject-bundled with curl download command and slow network
- (Reported by alex)
- [ASTERISK-27046] -
- 		res_pjsip_transport_websocket: segfault in get_write_timeout
- (Reported by Jørgen H)
- [ASTERISK-27039] -
- 		chan_pjsip: Device state is idle when channel from endpoint is in early media
- (Reported by Joshua Colp)
- [ASTERISK-26996] -
- 		chan_pjsip: Flipping between codecs
- (Reported by Michael Maier)
- [ASTERISK-26281] -
- 		chan_pjsip would send INVITE to 'Unreachable' endpoints
- (Reported by Jacek Konieczny)
- [ASTERISK-26973] -
- 		bridge: Crash when freeing frame and snooping
- (Reported by Michel R. Vaillancourt)
- [ASTERISK-19291] -
- 		Background in realtime
- (Reported by Andrew Nowrot)
- [ASTERISK-27025] -
- 		channel / meetme: Fix missing parentheses
- (Reported by Joshua Colp)
- [ASTERISK-27021] -
- 		GET /recordings/stored returns 500 Internal Server Error
- (Reported by Tim Morgan)
- [ASTERISK-24858] -
- 		[patch]Asterisk 13 PJSIP sends RTP packets in wrong byte order on Intel platform when using slin codec
- (Reported by Frankie Chin)
- [ASTERISK-23951] -
- 		Asterisk attempts and fails to build format_mp3 even if mp3lib was not downloaded
- (Reported by Tzafrir Cohen)
- [ASTERISK-25294] -
- 		srtp's crypto_get_random deprecated
- (Reported by Tzafrir Cohen)
- [ASTERISK-23839] -
- 		AGI - RECORD FILE - documentation doesn't describe BEEP argument
- (Reported by Rusty Newton)
- [ASTERISK-22432] -
- 		Async AGI crashes Asterisk when issuing "set variable" command without args
- (Reported by Antoine Pitrou)
- [ASTERISK-25662] -
- 		Malformed AGI 520 Usage response
- (Reported by Tony Mountifield)
- [ASTERISK-27008] -
- 		res_format_attr_h264: SDP parse fails if fmtp optional parameters have a space
- (Reported by John Harris)
- [ASTERISK-26399] -
- 		app_queue: Agent not called when caller is parked
- (Reported by wushumasters)
- [ASTERISK-26400] -
- 		app_queue: Queue member stops being called after AMI "Redirect" action for queues with wrapuptime
- (Reported by Etienne Lessard)
- [ASTERISK-26715] -
- 		app_queue: Member will not receive any new calls after doing a transfer if wrapuptime = greater than 0 and using Local channel
- (Reported by David Brillert)
- [ASTERISK-26975] -
- 		app_queue: Non-zero wrapup time can cause agents not to receive queue calls after transfer queue call
- (Reported by Lorne Gaetz)
- [ASTERISK-27012] -
- 		app_confbridge: ConfBridge sometimes does not play user name recording while leaving
- (Reported by Robert Mordec)
- [ASTERISK-26979] -
- 		res_rtp_asterisk: SRTP unprotect failed with authentication failure 10 or 110
- (Reported by Javier Riveros )
- [ASTERISK-26982] -
- 		chan_sip: rtcp_mux setting may cause ice completion failure/delay if client offers rtcp-mux as negotiable
- (Reported by Stefan Engström)
- [ASTERISK-26939] -
- 		Out of bound memory access in PJSIP multipart parser crashes Asterisk
- (Reported by Sandro Gauci)
- [ASTERISK-26940] -
- 		Asterisk Skinny memory exhaustion vulnerability leads to DoS
- (Reported by Sandro Gauci)
- [ASTERISK-26938] -
- 		Heap overflow in CSEQ header parsing affects Asterisk chan_pjsip and PJSIP
- (Reported by Sandro Gauci)
- [ASTERISK-26789] -
- 		Audit manipulation of channel flags without locks
- (Reported by Joshua Colp)
- [ASTERISK-26998] -
- 		res_pjsip_session: INVITE retransmissions could still setup the same call again.
- (Reported by Richard Mudgett)
- [ASTERISK-26143] -
- 		res_rtp_asterisk: One way audio when transcoding
- (Reported by Henning Holtschneider)
- [ASTERISK-26333] -
- 		Problems with Blind Transfer, PJSIP (Aastra 6869i)
- (Reported by Matthias Binder)
- [ASTERISK-26606] -
- 		tcptls: Incorrect OpenSSL function call leads to misleading error report
- (Reported by Bob Ham)
- [ASTERISK-26983] -
- 		Crash in Manager Reload when TLS Config Changes
- (Reported by Joshua Elson)
- [ASTERISK-25032] -
- 		[patch]cel_odbc sometimes inserts CEL with wrong eventtime
- (Reported by Etienne Lessard)
- [ASTERISK-26173] -
- 		func_cdr: CDR function does not permit empty values to be assigned
- (Reported by gkloepfer)
- [ASTERISK-25506] -
- 		[patch]CONFBRIDGE failure after an app_confbrige.so module reload results in segfault or error/warning messages.
- (Reported by Frederic LE FOLL)
- [ASTERISK-24529] -
- 		Using AMI Action Bridge to on an already bridged channel causes the incorrect return priority to be used
- (Reported by Corey Farrell)
- [ASTERISK-26966] -
- 		bridge_simple: Add support for streams
- (Reported by Kevin Harwell)
- [ASTERISK-26974] -
- 		res_pjsip: Deadlock in T.38 framehook
- (Reported by Richard Mudgett)
- [ASTERISK-26908] -
- 		res_pjsip: The ChanIsAvail causes a res_pjsip session to be leaked.
- (Reported by Richard Mudgett)
- [ASTERISK-26959] -
- 		dial: Allow topology of dialing channel to influence dialed channel
- (Reported by Joshua Colp)
- [ASTERISK-25823] -
- 		SIGSEGV, Segmentation fault. - ../sysdeps/x86_64/strlen.S: No such file or directory.
- (Reported by Andreas Krüger)
- [ASTERISK-26926] -
- 		func_speex: Crash caused by frame with no datalen
- (Reported by Richard Kenner)
- [ASTERISK-26964] -
- 		res_pjsip_session: Wrong From on reinvite when request and To URI differ
- (Reported by Yasin CANER)
- [ASTERISK-26930] -
- 		pjproject/Makefile.rules for pjsip 2.6 build fails for non-SSE2 instrunction Linux
- (Reported by abelbeck)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-26929] -
- 		pjsip: Add database tables for RLS
- (Reported by Joshua Colp)
- [ASTERISK-26949] -
- 		sdp: Implement T.38
- (Reported by Joshua Colp)
- [ASTERISK-26953] -
- 		Asterisk crash if hep.conf have some missing parameters
- (Reported by Joel Vandal)
- [ASTERISK-26890] -
- 		STUN server with non-default-route transport causes INVITE delay
- (Reported by George Joseph)
- [ASTERISK-26951] -
- 		chan_sip: ACK with SDP does not update a direct media bridge
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26692] -
- 		res_rtp_asterisk: Crash in dtls_srtp_handle_timeout at res_rtp_asterisk (using chan_sip)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26835] -
- 		res_rtp_asterisk: Crash when freeing RTCP address string
- (Reported by Niklas Larsson)
- [ASTERISK-26853] -
- 		res_rtp_asterisk: Crash in pjnath when receiving packet
- (Reported by Adagio)
- [ASTERISK-26613] -
- 		format_wav: wav16 format read file only by 320 - half of frame
- (Reported by Vitaly K)
- [ASTERISK-26169] -
- 		format_ogg_vorbis: Memory leak using OGG in MixMonitor
- (Reported by Ivan Myalkin)
- [ASTERISK-21856] -
- 		STUN never works when asterisk started without internet access
- (Reported by Jeremy Kister)
- [ASTERISK-20984] -
- 		Audible clicks when playing sox encoded au file with STREAM FILE AGI command
- (Reported by Roman S.)
- [ASTERISK-26528] -
- 		[UBSAN] strings.h:signed integer overflow in ast_str_case_hash
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26851] -
- 		res_pjsip_sdp_rtp: RTP instance does not use same IP as explicit transport
- (Reported by Richard Begg)
- [ASTERISK-26903] -
- 		Listening TCP/TLS sockets stop when temporarily out of open files
- (Reported by Walter Doekes)
- [ASTERISK-26928] -
- 		pjsip: Add database tables for PUBLISH support
- (Reported by Joshua Colp)
- [ASTERISK-26927] -
- 		pjproject_bundled: Crash on pj_ssl_get_info() while ioqueue_on_read_complete().
- (Reported by Alexander Traud)
- [ASTERISK-26905] -
- 		pjproject_bundled: Merge 3 upstream deadlock patches into bundled
- (Reported by Ross Beer)
- [ASTERISK-26920] -
- 		app_queue: PAUSEALL/UNPAUSEALL does not log reason
- (Reported by Troy Bowman)
- [ASTERISK-26897] -
- 		chan_sip: Security vulnerability with client code header
- (Reported by Alex Villacís Lasso)
- [ASTERISK-25974] -
- 		Unused realtime MOH classes not purged on 'moh reload'
- (Reported by Sébastien Couture)
- [ASTERISK-26916] -
- 		res_pjsip: Excessive refcount reached on transport ao2 object
- (Reported by Ross Beer)
- [ASTERISK-21721] -
- 		SIP Failed to parse multiple Supported: headers
- (Reported by Olle Johansson)
- [ASTERISK-26915] -
- 		chan_sip: Session Timers required but refused wrongly.
- (Reported by Alexander Traud)
- [ASTERISK-26363] -
- 		res_pjsip: Bye sent to sip trunk is not authenticated even after receiving a 407 error code
- (Reported by Yaacov Akiba Slama)
- [ASTERISK-26896] -
- 		Overflow of buffer to PQEscapeStringConn with large app_args causes ABRT
- (Reported by twisted)
- [ASTERISK-26705] -
- 		libasteriskssl.so not found when asterisk is installed for the 1st time
- (Reported by George Joseph)
- [ASTERISK-26900] -
- 		sdp: Add support for connection address management and topology updating
- (Reported by Joshua Colp)
- [ASTERISK-21009] -
- 		xmpp_pubsub_unsubscribe: Could not create IQ when creating pubsub unsubscription on client
- (Reported by Marcello Ceschia)
- [ASTERISK-25490] -
- 		[patch]SDP crypto tag is validated incorrectly
- (Reported by Joerg Sonnenberger)
- [ASTERISK-26885] -
- 		channel: Support dynamic number of file descriptors
- (Reported by Joshua Colp)
- [ASTERISK-26086] -
- 		res_musiconhold: format option is not documented adequately
- (Reported by Jens Bürger)
- [ASTERISK-23996] -
- 		No core dumps because of res_musiconhold chdir.
- (Reported by Walter Doekes)
- [ASTERISK-24712] -
- 		xmpp: starttls problem causes connection spew
- (Reported by Matthias Urlichs)
- [ASTERISK-26814] -
- 		pjproject_bundled build fails to download pjproject source when using cURL
- (Reported by Gergely Dömsödi)
- [ASTERISK-23510] -
- 		JABBER_STATUS fails with improper code 7 for unavailable clients
- (Reported by Anthony Critelli)
- [ASTERISK-21855] -
- 		Asterisk crashes when XMPP message is sent (JabberSend) and no internet connection is available
- (Reported by Jeremy Kister)
- [ASTERISK-25622] -
- 		WARNING for "JABBER: socket read error" should be more specific
- (Reported by Sean Darcy)
- [ASTERISK-26515] -
- 		rtp_engine: Allocate RTP payloads on a per-session basis
- (Reported by Joshua Colp)
- [ASTERISK-26818] -
- 		cdr: Problem setting variables in h exten
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26850] -
- 		res_hep_pjsip: Asterisk insert wrong protocol name in "Protocol ID" field in HEP packets
- (Reported by Max Norba)
- [ASTERISK-26484] -
- 		res_pjsip_messaging: Crash when using invalid URI in MessageSend 'from' argument.
- (Reported by Vinod Dharashive)
- [ASTERISK-26776] -
- 		res_pjsip_pubsub: Crash when generating xpidf content
- (Reported by Andrew Green)
- [ASTERISK-26880] -
- 		Asterisk crashes when multiple speex users join confbridge with pp_vad and dtx enabled
- (Reported by Kirsty Tyerman)
- [ASTERISK-26875] -
- 		app_mixmonitor: Recording out of sync when 183 but no RTP
- (Reported by Aaron An)
- [ASTERISK-26862] -
- 		app_queue: Queue stops calling members with local interface after forwarding in previous call
- (Reported by Robert Mordec)
- [ASTERISK-26732] -
- 		res_rtp_asterisk: Implement RTCP Multiplexing - breaking WebRTC in Chrome
- (Reported by Dan Jenkins)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-26867] -
- 		autochan: Locking in a function ast_autochan_destroy() on destroyed channel (after masquerade).
- (Reported by Krzysztof Trempala)
- [ASTERISK-26869] -
- 		res_pjsip_refer: blind call transfer w/o a user name doesn't go to the s extension
- (Reported by Torrey Searle)
- [ASTERISK-26668] -
- 		core: Malformed pattern matching extension (various factors) results in crash
- (Reported by xrobau)
- [ASTERISK-26865] -
- 		chan_iax2: Reload of iax peer results in loss of host address/port
- (Reported by Richard Begg)
- [ASTERISK-26872] -
- 		Bundled pjproject fails to build when tarball downloaded with curl due to md5 verification failure in Docker containers (or when there is no terminal)
- (Reported by Matt Jordan)
- [ASTERISK-26717] -
- 		Document the fact that Asterisk HEP support only works with the PJSIP channel driver
- (Reported by Olivier Krief)
- [ASTERISK-26643] -
- 		Extra new line in Device field of DeviceStateChange AMI Event after restart of Asterisk
- (Reported by Roman Bedros)
- [ASTERISK-25237] -
- 		stasis_cache.c:845 caching_topic_exec: - misleading ERROR message
- (Reported by Smirnov Aleksey)
- [ASTERISK-26857] -
- 		chan_pjsip: Dialplan function race condition
- (Reported by Joshua Colp)
- [ASTERISK-26822] -
- 		pjsip/cli_commands: pjsip show channelstats shows wrong codec
- (Reported by Kevin Harwell)
- [ASTERISK-26353] -
- 		res_musiconhold: musiconhold seems to think that the general section is a class and issues warning
- (Reported by Jonathan Harris)
- [ASTERISK-26685] -
- 		res_pjsip: Crash when using IPv6 and Transport ws,wss
- (Reported by Michael Balen)
- [ASTERISK-24562] -
- 		app_voicemail: Cannot set fromstring on a per-mailbox basis
- (Reported by Mark Scholten)
- [ASTERISK-26842] -
- 		Websocket becomes disconnected when trying to place call from browser
- (Reported by Mark Michelson)
- [ASTERISK-26841] -
- 		chan_sip: Call not cancelled after receiving a 422 response
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-26839] -
- 		core: Implement stream topology changing in channels
- (Reported by Joshua Colp)
- [ASTERISK-26598] -
- 		Saynumber is trying to get "and" from "digits/" subfolder
- (Reported by Jonathan Harris)
- [ASTERISK-17067] -
- 		Long lines in call files cause spurious syntax error
- (Reported by Dave Olszewski)
- [ASTERISK-26796] -
- 		res_pjsip_transport_websocket: Via header is 'WS' when it should be 'WSS'
- (Reported by Jørgen H)
- [ASTERISK-26816] -
- 		Implement ast_read_stream in channels
- (Reported by Joshua Colp)
- [ASTERISK-25628] -
- 		res_config_pgsql: should match the behavior of other drivers so that queue_log can disable adaptive logging
- (Reported by Dmitry Wagin)
- [ASTERISK-26774] -
- 		core: Playback URL fails after some time
- (Reported by Igor Gamayunov)
- [ASTERISK-26825] -
- 		pjsip.conf.sample: user_agent: still refers to branch 12
- (Reported by Tzafrir Cohen)
- [ASTERISK-26823] -
- 		PJSIP: Persistent subscriptions can cause FRACKs if endpoint does not exist
- (Reported by Mark Michelson)
- [ASTERISK-26623] -
- 		res_pjsip: Crash when calling PJSIPShowEndpoint
- (Reported by Jørgen H)
- [ASTERISK-26808] -
- 		res_pjsip_outbound_registration doesn't know about network change events
- (Reported by George Joseph)
- [ASTERISK-26781] -
- 		bridge: Passing the 'p' (play tone) flag to Bridge() application results in garbled audio
- (Reported by Sean Bright)
- [ASTERISK-26782] -
- 		res_pjsip: URI requirement for fields is not consistently documented and error does not provide indication
- (Reported by Peter Sokolov)
- [ASTERISK-26793] -
- 		Implement ast_write_stream in channels
- (Reported by George Joseph)
- [ASTERISK-26812] -
- 		[patch] Fix download_externals To Allow The Use Of curl Or wget
- (Reported by Michael L. Young)
- [ASTERISK-18271] -
- 		Pattern matching with res_config_mysql extensions does not behave as expected
- (Reported by Charlie Smurthwaite)
- [ASTERISK-26811] -
- 		stream: Add streams to "core show channel"
- (Reported by Joshua Colp)
- [ASTERISK-18731] -
- 		[patch] DUNDi weight parameter not processed correctly
- (Reported by Peter Racz)
- [ASTERISK-26799] -
- 		res_pjsip: Using an auth object for inbound and outbound authentication fails.
- (Reported by Richard Mudgett)
- [ASTERISK-26669] -
- 		PJSIP Segfault 13.13.1 (Bundled PJSIP)
- (Reported by Nic Colledge)
- [ASTERISK-26738] -
- 		Frequent segfaults since activation of DNS SRV, in pjsip_auth_clt_reinit_req at /pjsip/sip_auth_client.c, and pj_atomic_inc_and_get at pj/os_core_unix.c
- (Reported by Michael Maier)
- [ASTERISK-25893] -
- 		Function vmauthenticate accesses uninitialized memory
- (Reported by Filip Jenicek)
- [ASTERISK-26580] -
- 		[patch] Error during LDAP modify action when user unregisters
- (Reported by Nicholas John Koch)
- [ASTERISK-26802] -
- 		[patch] Integrity Check Of PJSIP Download Fails
- (Reported by Michael L. Young)
- [ASTERISK-15858] -
- 		[patch] Fix query with double backslash in string literals and stop log warnings
- (Reported by Humberto Figuera)
- [ASTERISK-26057] -
- 		res_config_sqlite3 uses incorrect query - unnecessary escape
- (Reported by Stepan)
- [ASTERISK-23457] -
- 		SQlite3: Realtime queue loading fails after PRAGMA query result
- (Reported by Scott Griepentrog)
- [ASTERISK-26794] -
- 		http: Crash on Reload Only in ast_tcptls_server_start
- (Reported by Joshua Elson)
- [ASTERISK-26714] -
- 		Phone default have not ringing on ARM
- (Reported by Igor Goncharovsky)
- [ASTERISK-26696] -
- 		pjsip_pubsub: PJSIP Subscription Persistence in AstDB Does not update on subscription refresh
- (Reported by Zach R)
- [ASTERISK-26756] -
- 		res_pjsip_mwi: Asterisk does not terminate MWI subscription
- (Reported by Carl Fortin)
- [ASTERISK-26790] -
- 		Implement stream topology (non-change request) API usage in channels
- (Reported by George Joseph)
- [ASTERISK-26723] -
- 		VoiceMailPlayMsg not playing messages via realtime
- (Reported by Ryan Rittgarn)
- [ASTERISK-18286] -
- 		[patch] 'Silence' is truncated in Record()
- (Reported by var)
- [ASTERISK-26775] -
- 		app_queue: reset abandoned in service level
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26786] -
- 		Implement ast_stream_topology API
- (Reported by George Joseph)
- [ASTERISK-26248] -
- 		chan_pjsip: Error when calling PJSIP client with domain specified
- (Reported by Norbert Varga)
- [ASTERISK-26788] -
- 		core: Protect flags during ast_waitfor
- (Reported by Joshua Colp)
- [ASTERISK-26115] -
- 		pbx: AMI Originate ignore "failed" extension on call failure
- (Reported by Nasir Iqbal)
- [ASTERISK-26773] -
- 		stream: Add basic API
- (Reported by Joshua Colp)
- [ASTERISK-26785] -
- 		configs/samples: The 'identify' entry is in the wrong section in sorcery.conf.sample
- (Reported by Torrey Searle)
- [ASTERISK-26772] -
- 		Crash in srv.c on startup with pjsip
- (Reported by nappsoft)
- [ASTERISK-26770] -
- 		res_stasis_device_state: Duplicate subscriptions when multiple received at same time
- (Reported by Joshua Colp)
- [ASTERISK-26767] -
- 		ARI channelvars cause memory leak
- (Reported by Sébastien Duthil)
- [ASTERISK-26716] -
- 		ari: Channels with pre-dial handlers cannot be hung up via ARI
- (Reported by Tom Pawelek)
- [ASTERISK-26632] -
- 		core: Possibility of a frame "imbalance" leading to stuck channels.
- (Reported by Mark Michelson)
- [ASTERISK-25951] -
- 		res_agi: run_agi eats frames it shouldn't
- (Reported by George Joseph)
- [ASTERISK-26343] -
- 		ASTERISK-25951 causes issues for callerid manipulation through agi
- (Reported by Morten Tryfoss)
- [ASTERISK-26704] -
- 		res_odbc.conf contains deprecated configuration: 'pooling', 'shared_connections', 'limit', and 'idlecheck' options were replaced by 'max_connections'.
- (Reported by Anthony Messina)
- [ASTERISK-26765] -
- 		res_resolver_unbound: FRACK! Excessive ref count trap tripped.
- (Reported by Richard Mudgett)
- [ASTERISK-21094] -
- 		MixMonitorMute mutes through stream if already slinear (e.g. Originate)
- (Reported by David Woolley)
- [ASTERISK-26679] -
- 		Crash on invalid contact domain (pjsip aor)
- (Reported by Dmitriy)
- [ASTERISK-26699] -
- 		res_pjsip: Assertion when sending OPTIONS request to endpoint
- (Reported by Ross Beer)
- [ASTERISK-26754] -
- 		build_tools: make_build_h does not handle \ in user name
- (Reported by Kirill Katsnelson)
- [ASTERISK-26755] -
- 		app_queue: Random queues disappear on "core reload queue all"
- (Reported by Kirill Katsnelson)
- [ASTERISK-26735] -
- 		res_pjsip_endpoint_identifier_ip: "srv_lookups" after match in .conf has no effect
- (Reported by Michael Maier)
- [ASTERISK-26693] -
- 		res_pjsip_endpoint_identifier_ip: Add support for SRV
- (Reported by Joshua Colp)
- [ASTERISK-26743] -
- 		PJPROJECT: Detecting compiled max log level does not work.
- (Reported by Richard Mudgett)
- [ASTERISK-26731] -
- 		res_sorcery_memory_cache: memory leak on every sorcery memory cache populate
- (Reported by Ustinov Artem)
- [ASTERISK-26739] -
- 		voicemail API test: confuses expected and actual values
- (Reported by Tzafrir Cohen)
- [ASTERISK-26740] -
- 		voicemail API test: uses varlibdir instead of datadir for a sound file
- (Reported by Tzafrir Cohen)
- [ASTERISK-26665] -
- 		app_queue: Agent ringing, Caller hangup before timeout, no agent name logged - missing RINGNOANSWER?
- (Reported by Marek Cervenka)
- [ASTERISK-26710] -
- 		[patch] res_rtp_asterisk: CHANNEL arguments, (rtcp,all_rtt),(rtcp,all_loss),(rtcp,all_jitter) always return 0
- (Reported by Aaron An)
- [ASTERISK-26672] -
- 		Crash when setting remote address on RTP instance
- (Reported by Richard Mudgett)
- [ASTERISK-26670] -
- 		[patch] Outgoing SIP-URI Dialing via PJSIP
- (Reported by Alexander Traud)
- [ASTERISK-26691] -
- 		Remember SDP negotiation on SIP_CODEC_INBOUND.
- (Reported by Alexander Traud)
- [ASTERISK-26673] -
- 		chan_pjsip: Crash when using CHANNEL dialplan function around masquerade
- (Reported by Joshua Colp)
- [ASTERISK-26684] -
- 		res_pjsip: Various issues with compact SIP headers
- (Reported by Joshua Elson)
- [ASTERISK-26683] -
- 		res_calendar: Calendars duplicated after module reload
- (Reported by Martin Tomec)
- [ASTERISK-26655] -
- 		[patch]pjsip: Transfers Broken with Compact Headers Enabled
- (Reported by JoshE)
- [ASTERISK-26621] -
- 		app_queue: Queue application does not ring members with Local interface
- (Reported by Jonas Kellens)
- [ASTERISK-26586] -
- 		chan_sip: Segfaults upon reload if client with MWI wasn't registered
- (Reported by Michael Kuron)
- [ASTERISK-25494] -
- 		build: GCC 5.1.x catches some new const, array bounds and missing paren issues
- (Reported by George Joseph)
- [ASTERISK-24499] -
- 		Need more explicit debug when PJSIP dialstring is invalid
- (Reported by Rusty Newton)
- [ASTERISK-25083] -
- 		Message.c: Message channel becomes saturated with frames leading to spammy log messages
- (Reported by Jonathan Rose)
- [ASTERISK-26653] -
- 		pjproject_bundled doesn't verify already downloaded tarballs
- (Reported by George Joseph)
- [ASTERISK-26433] -
- 		chan_sip: Allows To-tag checks to be bypassed, setting up new calls
- (Reported by Walter Doekes)
- [ASTERISK-26579] -
- 		codec_opus: Recursiveness when parsing fmtp line
- (Reported by Jørgen H)
- [ASTERISK-26644] -
- 		PJSIPShowRegistrationsInbound just dumps all aors
- (Reported by George Joseph)
- [ASTERISK-26647] -
- 		Support older DNS style for OpenBSD
- (Reported by snuffy)
- [ASTERISK-26490] -
- 		res_pjsip: sends 481 Call/Transaction Does Not Exist when transaction branch parameter contains "_"
- (Reported by Juris Breicis)
- [ASTERISK-26629] -
- 		tests/manager: 4 test failures as a result of iostream change
- (Reported by Joshua Colp)
- [ASTERISK-26109] -
- 		Asterisk fails building with OpenSSL 1.1.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-26617] -
- 		res_rtp_asterisk: Can't bind on systems without IPv6
- (Reported by Guido Falsi)
- [ASTERISK-26603] -
- 		[patch] chan_pjsip: not switching sending codec to receiving codec when asymmetric_rtp_codec=no
- (Reported by Alexei Gradinari)
- [ASTERISK-24330] -
- 		Requirement for 'wss' value in Contact header transport parameter on inbound traffic violates RFC7118
- (Reported by Marek Cervenka)
- [ASTERISK-26566] -
- 		res_rtp_asterisk: RTT miscalculation in RTCP
- (Reported by Hector Royo Concepcion)
- [ASTERISK-26604] -
- 		chan_sip: sip reload doesn't apply changes to tlscertfile, tlsciphers, etc.
- (Reported by Michael Kuron)
- [ASTERISK-26608] -
- 		Compile and link failures on OpenBSD
- (Reported by snuffy)
- [ASTERISK-26520] -
- 		codec_opus: Generated fmtp line has no content
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26605] -
- 		codec_opus: Spammed warning when Opus negotiated but codec_opus not loaded.
- (Reported by Richard Mudgett)
- [ASTERISK-26516] -
- 		pjsip: Memory corruption with possible memory leak.
- (Reported by Richard Mudgett)
- [ASTERISK-24515] -
- 		Unconditional use of fopencookie() / funopen() is non-portable
- (Reported by Timo Teräs)
- [ASTERISK-26556] -
- 		manager: AMI version report same in Ast 13 &amp; 14, despite Ast 14 syntax changes
- (Reported by Michelle Dupuis)
- [ASTERISK-26592] -
- 		Latest libedit (3.1) defaults to unicode and makes asterisk CLI read garbage
- (Reported by George Joseph)
- [ASTERISK-26575] -
- 		testsuite: Need to check PJSIP functionality when res_srtp is not loaded.
- (Reported by Joshua Colp)
- [ASTERISK-26565] -
- 		chan_unistim on 11, 13, 14 placing call on hold temporarily locks up set
- (Reported by Jason)
- [ASTERISK-26573] -
- 		Some typos in documentation of chan_sip.c
- (Reported by C.J. Collier)
- [ASTERISK-26571] -
- 		res_pjsip: Resolution incorrect when explicit IPv6 transport configured
- (Reported by Joshua Colp)
- [ASTERISK-26468] -
- 		ari: Bridge events stop working after this sequence of ARI calls
- (Reported by Daniele Pallastrelli)
- [ASTERISK-24400] -
- 		ooh323 sends wrong hangup code
- (Reported by Dmitry Melekhov)
- [ASTERISK-26555] -
- 		Multi-party Video: Fix some post Asterisk-11 regressions
- (Reported by Matt Jordan)
- [ASTERISK-26412] -
- 		build: Prepare for gcc 6.2
- (Reported by George Joseph)
- [ASTERISK-26509] -
- 		A few non-critical deprecation warnings when building on Ubuntu 16.10
- (Reported by Jonathan Harris)
- [ASTERISK-26523] -
- 		chan_sip: Asterisk 13.12.1 disconnects incoming calls after 2 minutes - rtptimeout behaving badly - regression
- (Reported by Michael Keuter)
- [ASTERISK-26549] -
- 		app_dial: When PickupChan() is used some channels may have incorrect device state
- (Reported by Joshua Colp)
- [ASTERISK-24274] -
- 		[patch]Codec Format Is Not Included in the SDP Media Attributes When SLIN48 Codec Is Used
- (Reported by Frankie Chin)
- [ASTERISK-26311] -
- 		[patch] rtp_engine: Allow more than 32 dynamic payload types.
- (Reported by Alexander Traud)
- [ASTERISK-26546] -
- 		mips64el and x32 - undefined reference to symbol 'dlopen@@GLIBC_2.2'
- (Reported by Tzafrir Cohen)
- [ASTERISK-26541] -
- 		res_pjsip_sdp_rtp: Restrict number of formats to maximum
- (Reported by Joshua Colp)
- [ASTERISK-26476] -
- 		chan_sip: Incorrect display option "Outbound reg. retry 403" in "sip show settings"
- (Reported by Sergey Grachev)
- [ASTERISK-25070] -
- 		Fix FTBFS on Hurd
- (Reported by Gabriele Giacone)
- [ASTERISK-26537] -
- 		AMI: NewConnectedLine event is not documented
- (Reported by Etienne Lessard)
- [ASTERISK-26526] -
- 		[UBSAN] vector.h: null pointer can be passed as argument 2 to memcpy
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26524] -
- 		astobj2: data_size variable is wasted space when AO2_DEBUG is not enabled.
- (Reported by Corey Farrell)
- [ASTERISK-26344] -
- 		Asterisk 13.11.0 + PJSIP crash
- (Reported by Ian Gilmour)
- [ASTERISK-26387] -
- 		Asterisk segfaults shortly after starting even with no active calls.
- (Reported by Harley Peters)
- [ASTERISK-26506] -
- 		[patch]res_pjsip_outbound_publish: Crash when publishing, in publisher_client_send at res_pjsip_outbound_publish.c
- (Reported by Matt Krokosz)
- [ASTERISK-26513] -
- 		tests/channels/pjsip/qualify/auth: Crashing enough to be a nuisance
- (Reported by Joshua Colp)
- [ASTERISK-26514] -
- 		Super Awesome Company: Don't specify transport in pjsip.conf
- (Reported by Rusty Newton)
- [ASTERISK-26510] -
- 		pjproject_bundled uses the --strip-components option of tar which isn't supported in older versions
- (Reported by George Joseph)
- [ASTERISK-22480] -
- 		Embedded pjproject: build.mak contains hardcoded full path to version.mak
- (Reported by Matt Jordan)
- [ASTERISK-26480] -
- 		[patch] CLI: core set debug: Auto-completes File not Module
- (Reported by Alexander Traud)
- [ASTERISK-26307] -
- 		res_pjsip_caller_id: Crash on outgoing change
- (Reported by Bill Brigden)
- [ASTERISK-26503] -
- 		app_voicemail: Asterisk crashes when MailboxExists is used
- (Reported by Doug Lytle)
- [ASTERISK-26423] -
- 		res_pjsip_sdp_rtp: Asymmetric RTP codec can cause audio loss and wonkiness
- (Reported by Andreas Wetzel)
- [ASTERISK-26309] -
- 		[patch] res_pjsip: Allow IPv4/IPv6 (Dual Stack) installations.
- (Reported by Alexander Traud)
- [ASTERISK-26482] -
- 		[patch] chan_pjsip: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-26455] -
- 		cdr_radius / cel_radius: try fix memory leak
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26421] -
- 		Segmentation Fault with ARI originate into mixing bridge with 43 clients
- (Reported by Andrew Nagy)
- [ASTERISK-26444] -
- 		'features show' command in CLI does not return prompt.
- (Reported by John Kiniston)
- [ASTERISK-26356] -
- 		menuselect: invalid test for GTK2
- (Reported by Tzafrir Cohen)
- [ASTERISK-26477] -
- 		pjproject: SEGV during SSL operations
- (Reported by George Joseph)
- [ASTERISK-26462] -
- 		[patch] app_queue: While using queues with realtime, setting back to an empty context doesn't stop the exit key usage
- (Reported by Leandro Dardini)
- [ASTERISK-26439] -
- 		chan_rtp: Crash when originating
- (Reported by Kayode)
- [ASTERISK-17470] -
- 		[patch] - When videosupport=yes, asterisk allows one end peer to send video, even though the other end supports only audio.
- (Reported by effie mouzeli)
- [ASTERISK-26416] -
- 		pjproject-bundled: configure fails to check for all required utilities
- (Reported by Corey Farrell)
- [ASTERISK-26466] -
- 		core: Be forgiving on external callerid that may be flawed so we don't drop events
- (Reported by Richard Mudgett)
- [ASTERISK-26362] -
- 		res_config_mysql: Broken after 13.10
- (Reported by Carlos Chavez)
- [ASTERISK-26446] -
- 		app_dial: There's no way to override the hangupcause on unanswered channels
- (Reported by George Joseph)
- [ASTERISK-26457] -
- 		[patch] force_rport,auto_comedia: No NAT detection triggered.
- (Reported by Alexander Traud)
- [ASTERISK-26453] -
- 		res_pjsip_config_wizard: Memory leak in module_unload
- (Reported by Badalian Vyacheslav)
- [ASTERISK-26410] -
- 		core: Asterisk 14 doesn't show the header in the console or verbose when starting
- (Reported by Dan Jenkins)
- [ASTERISK-24311] -
- 		Populating database via Alembic fails when using same database for multiple schema sets
- (Reported by Dafi Ni)
- [ASTERISK-26438] -
- 		[patch] chan_sip: auto_force_rport: No NAT = No Symmetric Response.
- (Reported by Alexander Traud)
- [ASTERISK-26330] -
- 		app_queue: Changing the "ringinuse" parameter of a queue doesn't affect dynamic members
- (Reported by Etienne Lessard)
- [ASTERISK-26426] -
- 		format_ogg_opus: remove from source
- (Reported by Kevin Harwell)
- [ASTERISK-18232] -
- 		Broken REGISTER sent to IPv4 server when bindaddr=[::]
- (Reported by Jacek)
- [ASTERISK-25468] -
- 		Deadlock in chan_sip - core show locks shows do_monitor lock
- (Reported by Barry Flanagan)
- [ASTERISK-26397] -
- 		manager: PresenceState action crashes Asterisk 14
- (Reported by Andrew Nagy)
- [ASTERISK-26389] -
- 		res_odbc: Clean up pooling options
- (Reported by Joshua Colp)
- [ASTERISK-26273] -
- 		core: Won't compile when LOW_MEMORY is enabled
- (Reported by Anthony Messina)
- [ASTERISK-26391] -
- 		Consoles do not display verbose logger messages even when requested.
- (Reported by Marcelo Terres)
- [ASTERISK-26352] -
- 		Astcanary dies when doing "core restart"
- (Reported by Walter Doekes)
- [ASTERISK-19867] -
- 		asterisk fails to lower its priority when astcanary dies
- (Reported by Xavier Hienne)
- [ASTERISK-26263] -
- 		SQL error when using realtime and registering extension / inserting into ps_contacts
- (Reported by Jeppe Ryskov Larsen)
- [ASTERISK-26365] -
- 		rtp: Offer with multiple payloads for same codec is incorrectly handled
- (Reported by Joshua Colp)
- [ASTERISK-26374] -
- 		res_pjsip_multihomed: Contact port is rewritten for connectionful protocols
- (Reported by Joshua Colp)
- [ASTERISK-26359] -
- 		[patch] cdr_mysql: fails to use UTC if so instructed
- (Reported by Tzafrir Cohen)
- [ASTERISK-26367] -
- 		rtp: Timestamps broken when video frame is across multiple RTP packets
- (Reported by Joshua Colp)
- [ASTERISK-26375] -
- 		res_pjsip_transport_management: Log message states seconds, but time value is milliseconds
- (Reported by Joshua Colp)
- [ASTERISK-19968] -
- 		TCP Session-Timers not dropping call
- (Reported by Aaron Hamstra)
- [ASTERISK-26364] -
- 		res_pjsip: Don't assume a request will have target addresses
- (Reported by Joshua Colp)
- [ASTERISK-26360] -
- 		app_queue: "queue show" output gets "failed to extend from 240 to 327" msgs.
- (Reported by Richard Mudgett)
- [ASTERISK-26358] -
- 		chan_sip: Contact is updated on re-200, but not on re-INVITE
- (Reported by Walter Doekes)
- [ASTERISK-26316] -
- 		res_pjsip_callerid: Irregular URI causes unexpected callerid
- (Reported by Kevin Harwell)
- [ASTERISK-26349] -
- 		13.11.1 res_pjsip/pjsip_distributor.c: Request 'REGISTER' failed
- (Reported by Dmitry Melekhov)
- [ASTERISK-26317] -
- 		res_pjsip_session: Add ability to use preferred codec only
- (Reported by Aaron An)
- [ASTERISK-26264] -
- 		res_pjsip: Crash when applying ACL from non-existent endpoint
- (Reported by nappsoft)
- [ASTERISK-26272] -
- 		chan_sip: File descriptors leak (UDP sockets)
- (Reported by Etienne Lessard)
- [ASTERISK-20234] -
- 		SRTP not working with some devices (Eg snom320) - Message "We are requesting SRTP for audio, but they responded without it!"
- (Reported by tootai)
- [ASTERISK-26341] -
- 		ARI: Stopping a media playlist only stops the current media URI being played back, and not the whole list
- (Reported by Matt Jordan)
- [ASTERISK-26291] -
- 		res_pjsip_session: segfault on already disconnected session
- (Reported by Alexei Gradinari)
- [ASTERISK-23989] -
- 		[patch]SDP offer/answer fails if crypto keys added to non-crypto offer
- (Reported by Olle Johansson)
- [ASTERISK-25691] -
- 		Crash occurs when screening mode (Dial's 'p' argument) is enabled and callee rejects a call or hangs up.
- (Reported by Etienne Lessard)
- [ASTERISK-26331] -
- 		Crash on “core show channeltype Surrogate” in ast_format_cap_get_names
- (Reported by CGI.NET)
- [ASTERISK-26085] -
- 		app_mp3: results in timeout for streams
- (Reported by Jens Bürger)
- [ASTERISK-26269] -
- 		res_pjsip: Wrong state for aors without registered contacts after startup
- (Reported by nappsoft)
- [ASTERISK-26226] -
- 		pbx: Asterisk crash on AMI action "ShowDialplan" when there's a circular dependency between contexts
- (Reported by Etienne Lessard)
- [ASTERISK-26299] -
- 		app_queue: Queue application sometimes stops calling members with Local interface
- (Reported by Etienne Lessard)
- [ASTERISK-26279] -
- 		pjproject-bundled: Fails to compile on Debian 6
- (Reported by George Joseph)
- [ASTERISK-26306] -
- 		channel: Hang-up crashes, chan_pjsip not cleaning up properly
- (Reported by Alexander Traud)
- [ASTERISK-26203] -
- 		res_fax: Deadlock when using FAXOPT(gateway)=yes with Local channels
- (Reported by Etienne Lessard)
- [ASTERISK-24822] -
- 		Deadlock: Fax Gateway framehook creates locking inversion in T.38 query option with features bridging code
- (Reported by David Brillert)
- [ASTERISK-22732] -
- 		Deadlock potential in res_fax and CCSS with local channels.
- (Reported by Richard Mudgett)
- [ASTERISK-26282] -
- 		AEL: macro-call in Dial application, macro "lacks 's' extension"
- (Reported by chris de rock)
- [ASTERISK-22820] -
- 		[patch] Plaintext auth is still supported in IAX2
- (Reported by Eugene)
- [ASTERISK-22374] -
- 		Finish mapping the sip.conf parameters to res_sip.conf parameters
- (Reported by Matt Jordan)
- [ASTERISK-24425] -
- 		[patch] jabber/xmpp to use TLS instead of SSLv3, security fix POODLE (CVE-2014-3566)
- (Reported by abelbeck)
- [ASTERISK-26228] -
- 		res_pjsip_sdp_rtp: G729A does not include annexb=no attribute.
- (Reported by Ali Ghavidel)
- [ASTERISK-25472] -
- 		Swagger scripts are not replacing format variable in file brief
- (Reported by Corey Farrell)
- [ASTERISK-25984] -
- 		res_odbc relies on res_odbc_transaction, but it's not mandatory to compile it
- (Reported by József Dudás)
- [ASTERISK-26305] -
- 		Asterisk 14: Two resolver unbound testsuite tests fail
- (Reported by Richard Mudgett)
- [ASTERISK-26288] -
- 		followme: fails to reset config items to default values on reload
- (Reported by Tzafrir Cohen)
- [ASTERISK-26303] -
- 		[patch] BuildSystem: ca_list_path capabilities not detected in PJProject.
- (Reported by Alexander Traud)
- [ASTERISK-25492] -
- 		ARI: Path parameters are case sensitive
- (Reported by Joshua Colp)
- [ASTERISK-26164] -
- 		XMPP no longer triggers NOTIFY to device via chan_pjsip
- (Reported by Ross Beer)
- [ASTERISK-26233] -
- 		pbx: Failure to remove inconsistent extension names
- (Reported by Corey Farrell)
- [ASTERISK-26246] -
- 		Security: Privilege escalation by AMI adding dialplan extensions.
- (Reported by Richard Mudgett)
- [ASTERISK-26267] -
- 		ast_register_atexit callbacks should be run on failed startup.
- (Reported by Corey Farrell)
- [ASTERISK-26241] -
- 		res_pjsip: When using compact headers, rpid and pai are incorrectly generated
- (Reported by George Joseph)
- [ASTERISK-25797] -
- 		app_queue: Crash when calling a queue with a member with a forward to an nonexistent extension
- (Reported by Etienne Lessard)
- [ASTERISK-26239] -
- 		res_pjsip_logger: An empty global/debug option is treated as a "match all" hostname
- (Reported by George Joseph)
- [ASTERISK-26238] -
- 		res_pjsip: Empty global default_from_user causes crash
- (Reported by Joshua Colp)
- [ASTERISK-26268] -
- 		alembic: 'auth_username' not in PJSIP 'identify_by' enum
- (Reported by Joshua Colp)
- [ASTERISK-26253] -
- 		sdp_srtp: libsrtp now a required dependency, shouldn't be
- (Reported by Ben Merrills)
- [ASTERISK-26145] -
- 		pjsip: Deadlock with suspend + masquerade + indicate
- (Reported by Ross Beer)
- [ASTERISK-26183] -
- 		alembic: error when using sqlalchemy version 1.1.0b2
- (Reported by Kevin Harwell)
- [ASTERISK-26283] -
- 		res_resolver_unbound: fails configure on older Ubuntu and CentOS
- (Reported by George Joseph)
- [ASTERISK-26280] -
- 		DNS lookups can block channel media paths
- (Reported by Mark Michelson)
- [ASTERISK-26278] -
- 		asterisk.h should produce a reasonable error for external modules that fail to define AST_MODULE_SELF_SYM.
- (Reported by Corey Farrell)
- [ASTERISK-25217] -
- 		[patch]res_pjsip_outbound_publish.c needs a similar treatment for module unloading as res_pjsip_outbound_registration.c
- (Reported by Richard Mudgett)
- [ASTERISK-26265] -
- 		Errors ignored from some parts of system initialization.
- (Reported by Corey Farrell)
- [ASTERISK-26206] -
- 		[patch] res_pjsip: Use more compatible regex for get all
- (Reported by Dmitry Wagin)
- [ASTERISK-26256] -
- 		[patch] SIP/SDP origin (o=) contains brackets with IP6
- (Reported by Alexander Traud)
- [ASTERISK-25996] -
- 		Remove "live_dangerously" requirement on DB(read)
- (Reported by Andrew Nagy)
- [ASTERISK-26148] -
- 		pjsip: Cannot compile 13.10.0-rc1: "libasteriskpj.so: undefined reference to..."
- (Reported by Hans van Eijsden)
- [ASTERISK-26237] -
- 		Fax is detected on regular calls.
- (Reported by Richard Mudgett)
- [ASTERISK-26227] -
- 		sqlalchemy error due to long identifier name
- (Reported by Mark Michelson)
- [ASTERISK-14] -
- 		asterisk leaves zombie mpg123
- (Reported by dcarr)
- [ASTERISK-23013] -
- 		[patch] Deadlock between 'sip show channels' command and attended transfer handling
- (Reported by Ben Smithurst)
- [ASTERISK-26199] -
- 		PJSIP: tx_data_destroy called twice
- (Reported by Scott Griepentrog)
- [ASTERISK-26166] -
- 		res_pjsip_pubsub: Crash when decrementing reference count of message
- (Reported by Ross Beer)
- [ASTERISK-26174] -
- 		res_pjsip: Crash when freeing cloned message in distributor
- (Reported by Ross Beer)
- [ASTERISK-26216] -
- 		res_fax: Deadlock when detect fax while channel executing Playback
- (Reported by Richard Mudgett)
- [ASTERISK-26214] -
- 		Allow arbitrary time for fax detection to end on a channel
- (Reported by Richard Mudgett)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27063] -
- 		Add support for systemd socket activation
- (Reported by Corey Farrell)
- [ASTERISK-27117] -
- 		core: Add support for timelen parsing to ast_parse_arg and ACO.
- (Reported by Corey Farrell)
- [ASTERISK-27129] -
- 		ast_waitfordigit_full: add support for filtering DTMF keys which can break the wait.
- (Reported by Corey Farrell)
- [ASTERISK-26995] -
- 		Add QUEUE_FLOAT_PENALTY to app_queue
- (Reported by Steve Davies)
- [ASTERISK-26878] -
- 		func_channel: Add ability to get the callid so dialplan has access to it.
- (Reported by Richard Mudgett)
- [ASTERISK-26863] -
- 		res_pjsip: Add endpoint identification scheme based on a configured SIP header/value
- (Reported by Matt Jordan)
- [ASTERISK-17428] -
- 		[patch] Allow "Comedian Mail" branding to be removed
- (Reported by John Covert)
- [ASTERISK-26584] -
- 		[patch] RTCP feedback for codec modules
- (Reported by Lorenzo Miniero)
- [ASTERISK-19862] -
- 		app_queue: Update Data of Queues (use queues as outbound calls container)
- (Reported by Sebastian Gutierrez)
- [ASTERISK-26630] -
- 		Make logging PJPROJECT messages a bit easier
- (Reported by Richard Mudgett)
- [ASTERISK-26587] -
- 		app_originate: Add option to execute gosub prior to dial
- (Reported by dkerr)
- [ASTERISK-26595] -
- 		ARI: Add the ability to control the source of video in a multi-party mixing bridge
- (Reported by Matt Jordan)
- [ASTERISK-26492] -
- 		ARI: Add ability to specify channel variables on websocket events
- (Reported by Mark Michelson)
- [ASTERISK-26470] -
- 		ARI: Add an 'asterisk_id' field to outgoing events
- (Reported by Matt Jordan)
- [ASTERISK-26277] -
- 		Add dialplan function PJSIP_SEND_SESSION_REFRESH that sends a session refresh to update formats on a channel after session establishment
- (Reported by Matt Jordan)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
</package>
<package pkgid="2d6789d617056c37ea35a7d83c178423ea9f4afd7de6a4656ca6b6150b1a6c2e" name="asterisk" arch="src">
  <version epoch="0" ver="15.2.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27278] -
- 		[patch] chan_sip: Provide access to read the full SIP Request-URI from INVITE
- (Reported by David J. Pryke)
- [ASTERISK-27255] -
- 		alembic: Add support for Microsoft SQL server
- (Reported by Florian Floimair)
- [ASTERISK-27253] -
- 		[patch] libsrtp-2.1.x support
- (Reported by Alexander Traud)
- [ASTERISK-27220] -
- 		Enable CHANNEL function to get from and to tag from SIP Headers
- (Reported by Andre Nazario)
- [ASTERISK-27169] -
- 		Google OAuth 2.0 support for XMPP / Motif
- (Reported by Andrey)
- [ASTERISK-27173] -
- 		Support for GMIME 3.0
- (Reported by Tzafrir Cohen)
- [ASTERISK-27085] -
- 		[patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
- (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27346] -
- 		res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
- (Reported by Ronald Raikes)
- [ASTERISK-27372] -
- 		ARI: Node ARI client broken in latest versions of 13 and 14
- (Reported by Benjamin Keith Ford)
- [ASTERISK-27047] -
- 		res_pjsip: user=phone added to Anonymous caller-id when it shouldn't be.
- (Reported by dtryba)
- [ASTERISK-26988] -
- 		res_pjsip_session: user_eq_phone adds double user=phone parameters to URIs
- (Reported by dtryba)
- [ASTERISK-27270] -
- 		cdr_mysql: various crashes at second module reload if cdr_mysql.conf is configured
- (Reported by Tzafrir Cohen)
- [ASTERISK-25266] -
- 		Application Originate returns SUCCESS to ORIGINATE_STATUS upon failure to originate
- (Reported by Allen Ford)
- [ASTERISK-27192] -
- 		res_pjsip: Loss of SIP registrations causing unavailable endpoints
- (Reported by Richard Mudgett)
- [ASTERISK-27305] -
- 		res_ari: Memory leaks in ARI when using Content-Type: application/json
- (Reported by David Hajek)
- [ASTERISK-26922] -
- 		chan_sip: tcpbind uses wrong source address
- (Reported by Ksenia)
- [ASTERISK-27324] -
- 		[patch] Dual-Stack server cannot be used as IPv4 client via TCP/TLS
- (Reported by Alexander Traud)
- [ASTERISK-27317] -
- 		vector: multiple evaluation of elem in AST_VECTOR_ADD_SORTED.
- (Reported by Corey Farrell)
- [ASTERISK-27301] -
- 		[patch] app_queue: Music On Hold for real-time queues is not reset to default
- (Reported by Nathan Bruning)
- [ASTERISK-27318] -
- 		res_pjsip_mwi: uninitialized value from ast_strings_match
- (Reported by Corey Farrell)
- [ASTERISK-27284] -
- 		Status of RFC 3323 and PJSIP
- (Reported by dtryba)
- [ASTERISK-27296] -
- 		[patch] False positive busy checks when icalendar's recurrence-id mechanism is involved
- (Reported by Benoît Dereck-Tricot)
- [ASTERISK-27216] -
- 		app_queue: does its check-makeannouncement-logic twice each head-caller-loop
- (Reported by Stefan Engström)
- [ASTERISK-27298] -
- 		Problem with expires on pjsip / outbound-publish
- (Reported by Cyrille Demaret)
- [ASTERISK-27295] -
- 		Contact is improperly translated after d178f497
- (Reported by Sean Bright)
- [ASTERISK-27292] -
- 		Multiple RTP Stream Created Breaking RFC2833 (SSRC Changes)
- (Reported by Ross Beer)
- [ASTERISK-27259] -
- 		chan_pjsip: Outgoing leg does not use all configured codecs, but subset based on caller
- (Reported by lvl)
- [ASTERISK-27289] -
- 		A codeblock that maintains a bug,but maybe the codeblock will never run
- (Reported by Huangyx)
- [ASTERISK-27277] -
- 		bridge: Renegotiate if source stream changes.
- (Reported by Joshua Colp)
- [ASTERISK-27283] -
- 		Realtime config fail with PostgreSQL version before 9.1
- (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27264] -
- 		res_pjsip_session: Crashes after sending PRACK and receiving 200 OK
- (Reported by Daniel Heckl)
- [ASTERISK-27260] -
- 		[pjsip] chan_pjsip_indicate: Don't know how to indicate condition 36
- (Reported by Daniel Heckl)
- [ASTERISK-27257] -
- 		bridge_native_rtp: half-way direct media when using early bridging
- (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] -
- 		SRTP unprotect: authentication failure when RTP sequence number switches from 65535 -&gt; 0
- (Reported by Marcello Ceschia)
- [ASTERISK-27274] -
- 		RTCP needs better packet validation to resist port scans.
- (Reported by Richard Mudgett)
- [ASTERISK-27252] -
- 		RTP: One way audio with direct media and strictrtp=yes.
- (Reported by Richard Mudgett)
- [ASTERISK-27279] -
- 		Crash in pubsub_on_rx_request NULL pointer - Possible PJSIP Vulnerability
- (Reported by Ross Beer)
- [ASTERISK-25524] -
- 		module reload res_calendar.so does not reload everything in calendar.conf
- (Reported by Jesper)
- [ASTERISK-24588] -
- 		res_calendar does not process CalDAV from Owncloud [fix included]
- (Reported by Stefan Gofferje)
- [ASTERISK-25523] -
- 		res_calendar: Warning about invalid channel value (for notification) occurs even when event has no notification configured.
- (Reported by Jesper)
- [ASTERISK-21399] -
- 		RTP Multicast of L16 (type 10): Asterisk and wireshark disagree
- (Reported by Tzafrir Cohen)
- [ASTERISK-27248] -
- 		[patch]external_media_address and external_signaling_address don't always honor localnet
- (Reported by Walter Doekes)
- [ASTERISK-24066] -
- 		res_smdi: convert to astobj2
- (Reported by Corey Farrell)
- [ASTERISK-27217] -
- 		chan_sip: Asterisk crashing when subscription doesn't get set
- (Reported by Bryan Walters)
- [ASTERISK-17540] -
- 		SDP origin attribute modified when issuing re-INVITE because of directmedia=yes
- (Reported by saghul)
- [ASTERISK-27165] -
- 		CDR: CDR(start,u) function won't work in cdr_custom config
- (Reported by Jacek Konieczny)
- [ASTERISK-27254] -
- 		alembic: prune_on_boot fix erroneous
- (Reported by Florian Floimair)
- [ASTERISK-27232] -
- 		When in queue on g722 with interruptions, music on hold can get stuck and no longer play
- (Reported by Jens T.)
- [ASTERISK-27024] -
- 		nat/external_media settings ignored in 14.4.1
- (Reported by Christopher van de Sande)
- [ASTERISK-26879] -
- 		PJSIP external_media_address ignored if no local_net options are provided
- (Reported by Matt Jordan)
- [ASTERISK-27236] -
- 		Segfault ast_channel_name (chan=0x0) at channel_internal_api.c:478 during T.38 Fax Receive
- (Reported by Ross Beer)
- [ASTERISK-27225] -
- 		Crash when freeing dtls_cfg-&gt;cafile
- (Reported by Richard Kenner)
- [ASTERISK-27177] -
- 		ooh323c: misleading indentation in addons/ooh323c/src/ooSocket.c
- (Reported by Tzafrir Cohen)
- [ASTERISK-27241] -
- 		libc segfault upon entry into app_directory
- (Reported by David Moore)
- [ASTERISK-27152] -
- 		Sending a "tel" uri in a From or To header in an unauthenticated message causes asterisk to crash
- (Reported by Ross Beer)
- [ASTERISK-27103] -
- 		core: ast_safe_system command injection possible.
- (Reported by Corey Farrell)
- [ASTERISK-27013] -
- 		res_rtp_asterisk: Media can be hijacked even with strict RTP enabled
- (Reported by Joshua Colp)
- [ASTERISK-27231] -
- 		res_rtp_asterisk: Allow remote SSRC to change due to renegotiation
- (Reported by Joshua Colp)
- [ASTERISK-26994] -
- 		Confbridge: CBAnn channels intermittently become stuck when caller hangs up before recording name
- (Reported by James Terhune)
- [ASTERISK-27222] -
- 		core: Don't queue up multiple video update frames.
- (Reported by Joshua Colp)
- [ASTERISK-20858] -
- 		app_minivm fails to clean up mkstemp files
- (Reported by Walter Doekes)
- [ASTERISK-16777] -
- 		several filename bugs in Record() application
- (Reported by klaus3000)
- [ASTERISK-27168] -
- 		alembic: PJSIP scripts are missing column dtls_fingerprint in ps_endpoints table
- (Reported by Florian Floimair)
- [ASTERISK-27209] -
- 		Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
- (Reported by Torrey Searle)
- [ASTERISK-19103] -
- 		When using realtime queues, function QUEUE_MEMBER_LIST() will return an error if no other app/function has loaded the queues first. This problem does not exist if queues.conf is used.
- (Reported by Jim Van Meggelen)
- [ASTERISK-21241] -
- 		When using voicemail as announce only (maxmsg=0), the star dtmf to enter the voicemail is not honored
- (Reported by Eelco Brolman)
- [ASTERISK-27212] -
- 		bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU
- (Reported by Richard Mudgett)
- [ASTERISK-27204] -
- 		[patch] app_queue: Wrong queue stat calculation
- (Reported by sungtae kim)
- [ASTERISK-27207] -
- 		XMPP OAuth not working due to inverted logic
- (Reported by Michael Kuron)
- [ASTERISK-27174] -
- 		res_calendar_icalendar: Recurring events not being loaded from Google calendar using ical
- (Reported by Mark Thompson)
- [ASTERISK-27202] -
- 		If wget is not installed and "or" is not available, external components (excluding pjsip) are not installed
- (Reported by Seán C. McCord)
- [ASTERISK-27200] -
- 		manager: hook event is not being raised
- (Reported by Kevin Harwell)
- [ASTERISK-27147] -
- 		Either asterisk or pjproject isn't re-using tcp connections (again)
- (Reported by George Joseph)
- [ASTERISK-27193] -
- 		IPv6 receive address in message doesn't include brackets
- (Reported by Scott Griepentrog)
- [ASTERISK-27158] -
- 		[patch] res_rtp_asterisk: RTCP statistics are not available when native bridge is used
- (Reported by Torrey Searle)
- [ASTERISK-26745] -
- 		Asymmetric codecs when asymmetric_rtp_codec=no
- (Reported by Jesse Ross)
- [ASTERISK-27189] -
- 		Make --with-pjproject-bundled the default for Asterisk 15
- (Reported by George Joseph)
- [ASTERISK-27110] -
- 		RTP session is not fully destroyed on channel hangup
- (Reported by Matt Jordan)
- [ASTERISK-27182] -
- 		bridge: Crash when mapping streams
- (Reported by Joshua Colp)
- [ASTERISK-27180] -
- 		channel: requester leaks joint_cap on success.
- (Reported by Corey Farrell)
- [ASTERISK-27179] -
- 		res_pjsip_session: Handling of 'msid' is incorrect
- (Reported by Kevin Harwell)
- [ASTERISK-27119] -
- 		res_pjsip: parse/add msid attribute when webrtc is enabled
- (Reported by Kevin Harwell)
- [ASTERISK-27171] -
- 		Asterisk 15.0.0-Beta1 does not compile
- (Reported by Ira Emus)
- [ASTERISK-26659] -
- 		res_pjsip: PJSIP presence - missing braces around the status element in XML
- (Reported by Abraham Liebsch)
- [ASTERISK-27156] -
- 		Asterisk won't compile on Fedora 26 with devmode enabled.
- (Reported by Corey Farrell)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27215] -
- 		[patch]AMI : Add CancelAtxfer Action
- (Reported by Thomas Sevestre)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758404">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758405">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758406">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e66c30d6e433caaf84e697f9b37b3260d93074306ed27416ed0ac603a11abad4" name="asterisk" arch="src">
  <version epoch="0" ver="15.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.1-1" date="1515758400">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15. The available
- security releases are released as versions 13.13-cert7, 13.18.1,
- 14.7.1 and 15.1.1.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases2
-
- The release of these versions resolves the following security
- vulnerabilities:
-
-     AST-2017-009: Buffer overflow in pjproject header parsing can
-     cause a crash in Asterisk
-     By carefully crafting invalid values in the Cseq and the Via
-     header port, pjproject’s packet parsing code can create strings
-     larger than the buffer allocated to hold them. This will usually
-     cause Asterisk to crash immediately. The packets do not have to
-     be authenticated.
-
-     AST-2017-010: Buffer overflow in CDR’s set user
-     No size checking is done when setting the user field for Party B
-     on a CDR. Thus, it is possible for someone to use an arbitrarily
-     large string and write past the end of the user field storage
-     buffer. The earlier AST-2017-001 advisory for the CDR user field
-     overflow was for the Party A buffer.
-
-     AST-2017-011: Memory leak in pjsip session resource
-     A memory leak occurs when an Asterisk pjsip session object is
-     created and that call gets rejected before the session itself is
-     fully established. When this happens the session object never
-     gets destroyed.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.12
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.1
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.11
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert7
-
- The security advisories are available at:
-  * https://downloads.asterisk.org/pub/security/AST-2017-009.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-010.pdf
-  * https://downloads.asterisk.org/pub/security/AST-2017-011.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758401">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758402">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758403">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758404">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758405">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7593e1c3994a6cd617295adbf1c7d8d86ef693b6f2fe980f8514749635fc17ce" name="asterisk" arch="src">
  <version epoch="0" ver="15.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.2-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.1.2.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.1.2 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.1.2
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758402">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758403">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758404">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8b18d3a503020af06b4c28cd29620196c8c41d0e0f7b7cc76453aa07a85acccc" name="asterisk" arch="src">
  <version epoch="0" ver="15.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.3-1" date="1515758400">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert8, 13.18.3,
- 14.7.3 and 15.1.3.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-013: DOS Vulnerability in Asterisk chan_skinny
-   If the chan_skinny (AKA SCCP protocol) channel driver is
-   flooded with certain requests it can cause the asterisk
-   process to use excessive amounts of virtual memory
-   eventually causing asterisk to stop processing requests of
-   any kind.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog=14.7.3
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.3
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert8
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-013.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758401">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758402">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758403">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="72ec84cb515c25f22a8d44fe00337110ee58543f16c1c13857a03e3a1d14b485" name="asterisk" arch="src">
  <version epoch="0" ver="15.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.4-1" date="1515758400">- The Asterisk Development Team has announced security releases for
- Certified Asterisk 13.13 and Asterisk 13, 14 and 15.  The available
- security releases are released as versions 13.13-cert9, 13.18.4,
- 14.7.4 and 15.1.4.
-
- These releases are available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The release of these versions resolves the following security
- vulnerabilities:
-
- * AST-2017-012: Remote Crash Vulnerability in RTCP Stack
-   If a compound RTCP packet is received containing more than
-   one report (for example a Receiver Report and a Sender
-   Report) the RTCP stack will incorrectly store report
-   information outside of allocated memory potentially causing
-   a crash.
-
- For a full list of changes in the current releases, please see the
- ChangeLogs:
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.4
- http://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.4
- http://downloads.asterisk.org/pub/telephony/certified-asterisk/ChangeLog-certified-13.13-cert9
-
- The security advisories are available at:
- http://downloads.asterisk.org/pub/security/AST-2017-012.html
- http://downloads.asterisk.org/pub/security/AST-2017-012.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758401">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758402">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c16f7d9de31f4a92afcfc5658cfc9f54b84619d6fc89757875c626f08341d304" name="asterisk" arch="src">
  <version epoch="0" ver="15.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.1.5-1" date="1515758400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.18.5, 14.7.5, 15.1.5 and 13.18-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2017-014: Crash in PJSIP resource when missing a contact header
-     A select set of SIP messages create a dialog in Asterisk. Those SIP messages
-     must contain a contact header. For those messages, if the header was not
-     present and using the PJSIP channel driver, it would cause Asterisk to crash.
-     The severity of this vulnerability is somewhat mitigated if authentication is
-     enabled. If authentication is enabled a user would have to first be authorized
-     before reaching the crash point.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.18.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.5
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.1.5
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert2
-
- The security advisory is available at:
- https://downloads.asterisk.org/pub/security/AST-2017-014.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758401">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4b782d84ef3ada87aa7e083657408f90f1766129bcf7125d00dcda0de89483e1" name="asterisk" arch="src">
  <version epoch="0" ver="15.6.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.0-1" date="1515758400">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-  * ASTERISK-27478 - PJSIP: Add CHANNEL(pjsip,request_uri) to get
-       incoming INVITE Request-URI.
-       (Reported by Richard Mudgett)
-  * ASTERISK-27413 - Add cache_media_frames debugging option.
-    
-       (Reported by Richard Mudgett)
-  * ASTERISK-27206 - res_pjsip: No mechanism exists to limit
-       endpoint identification to IP only
-       (Reported by Ben
-       Merrills)
-
- Bugs fixed in this release:
- -----------------------------------
-  * ASTERISK-27531 - Compiler optimizations can break module load
-       sequence.
-       (Reported by abelbeck)
-  * ASTERISK-27480 - Security: Authenticated SUBSCRIBE without
-       Contact crashes asterisk
-       (Reported by Ross Beer)
-  * ASTERISK-27299 - Asterisk Hangs with Bad file descriptor on
-       read()
-       (Reported by Abhay Gupta)
-  * ASTERISK-25079 - AMI bridge of channels results in MOH not
-       destroyed and robotic audio on one channel
-       (Reported by
-       Zane Conkle)
-  * ASTERISK-27495 - DNS: Unexpected rr_type can cause crash
-     
-       (Reported by Corey Farrell)
-  * ASTERISK-27490 - chan_console: 'set active' fails to work
-    
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-24756 - ConfBridge sound_muted does not work from
-       CLI or AMI
-       (Reported by Thomas Frederiksen)
-  * ASTERISK-25649 - Transfer application does not work with
-       Local channels - documentation misleading
-       (Reported by
-       Ivan Ullmann)
-  * ASTERISK-25869 - chan_sip: "rejected because extension not
-       found" should be logged as a security event
-       (Reported by
-       Brian J. Murrell)
-  * ASTERISK-27440 - Strictrtp has issues to qualify video rtp
-       streams
-       (Reported by Wim De Vlaminck)
-  * ASTERISK-24329 - Music On Hold announcement cuts intro of
-       music the first time it is played
-       (Reported by Thomas
-       Frederiksen)
-  * ASTERISK-19657 - Coverity Report: Fix issues for error type
-       CHAR_IO
-       (Reported by Matt Jordan)
-  * ASTERISK-27175 - iax.conf demo peer is invalid
-      
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27430 - README refers to security documents that do
-       not exist.
-       (Reported by Corey Farrell)
-  * ASTERISK-20281 - "core set verbose" behaves strangely, can't
-       alias it, cli.conf example broken
-       (Reported by Tim
-       Ringenbach at Asteria Solutions Group)
-  * ASTERISK-27382 - crash after an invalid rtcp packet from GT48
-       FXS gateway
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27429 - res_rtp_asterisk: Multiple reports in an
-       RTCP packet will write past where it should
-       (Reported by
-       Vitezslav Novy)
-  * ASTERISK-27408 - Identify causes and fix
-       pjsip/resolver/srv/failover/in_dialog/transport_tcp
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-18411 - Queue members with hints for state_interface
-       get stuck in "In Use" state.
-       (Reported by Steven T.
-       Wheeler)
-  * ASTERISK-26131 - chan_sip: Crash Asterisk (in
-       sip_request_call at chan_sip.c) by making a call to a single
-       character in a dot pattern match
-       (Reported by Dwayne
-       Hubbard)
-  * ASTERISK-27475 - codec_opus requires libcurl
-       (Reported
-       by Samuel For)
-  * ASTERISK-27467 - pjsip_options: qualify_frequency sometimes
-       not applied on reload
-       (Reported by John Bigelow)
-  * ASTERISK-27465 - CLI Completion Not Working
-       (Reported
-       by Ross Beer)
-  * ASTERISK-27460 - CDR: Deadlock using AMI Originate with
-       Variable CDR(amaflags)=...
-       (Reported by Richard Mudgett)
-  * ASTERISK-27453 - RTP: Blind transfer direct media scenario
-       results in one way audio.
-       (Reported by Richard Mudgett)
-  * ASTERISK-20643 - SIP ICE support - remove hardcoded
-       limitation on SDP size, make ICE support disabled by default in
-       SIP, maybe provide a better warning message
-       (Reported by
-       Roy)
-  * ASTERISK-26980 - pjsip: Clean up WebRTC disables
-      
-       (Reported by abelbeck)
-  * ASTERISK-27452 - Security: chan_skinny:  Memory exhaustion if
-       flooded with unauthenticated requests
-       (Reported by George
-       Joseph)
-  * ASTERISK-27454 - res_http_post: Don't require
-       GMIME_MAJOR_VERSION
-       (Reported by Joshua Colp)
-  * ASTERISK-23735 - Transcoding makes bad choice in high-rate
-       translations
-       (Reported by Richard Kenner)
-  * ASTERISK-27445 - ARI: Updating a bridge gives wrong error
-       message.
-       (Reported by Frank Durden)
-  * ASTERISK-24662 - [patch] column and row headers for Signed
-       Linear format variants in output of 'core show translation' are
-       ambiguous
-       (Reported by Rusty Newton)
-  * ASTERISK-27353 - H323 audio starts with a delay of 2
-       seconds.
-       (Reported by Marco Giordani)
-  * ASTERISK-27442 - pjsip: 183 without To tag does not negotiate
-       media
-       (Reported by Kevin Harwell)
-  * ASTERISK-27437 - [patch] ICE: server-reflexive candidates
-       (srflx) with Dual-Stack.
-       (Reported by Alexander Traud)
-  * ASTERISK-27434 - [patch] chan_sip/ICE: Square brackets around
-       IPv6 addresses.
-       (Reported by Alexander Traud)
-  * ASTERISK-27435 - [patch] configure:
-       pjsip_evsub_set_uas_timeout not found.
-       (Reported by
-       Alexander Traud)
-  * ASTERISK-27332 - Asterisk fails to configure on MacOS Sierra
-
-       (Reported by Ivan Larionov)
-  * ASTERISK-27431 - Asterisk fails to build when openssl headers
-       are not installed.
-       (Reported by Corey Farrell)
-  * ASTERISK-27421 - RTP source learning not working with devices
-       that have some clock issues
-       (Reported by nappsoft)
-  * ASTERISK-27361 - Attended transfer crashes in Asterisk
-       13.17.2
-       (Reported by Alessandro Pimenta)
-  * ASTERISK-27238 - Bridging: Crash freeing a frame that's
-       already been freed
-       (Reported by Richard Kenner)
-  * ASTERISK-27412 - core: Audiohook freeing interpolated frame
-       when it shouldn't.
-       (Reported by Mikhail)
-  * ASTERISK-27423 - app_record:  We set the RECORD_STATUS
-       channel variable before closing the file
-       (Reported by
-       George Joseph)
-  * ASTERISK-26758 - res_hep_pjsip: For WebRTC clients Asterisk
-       insert same ip address in "source ip address" and "destination
-       ip address" fields in HEP packets
-       (Reported by Max Norba)
-  * ASTERISK-27363 - res_http_websocket: Wrong LocalAddress (it
-       is equal to RemoteAddress)
-       (Reported by Vasilii Rogin)
-  * ASTERISK-27415 - asterisk.conf: Setting astctl without
-       setting astrundir is ineffective.
-       (Reported by Corey
-       Farrell)
-  * ASTERISK-27411 - pjsip: TCP connections may not be destroyed
-
-       (Reported by Joshua Colp)
-  * ASTERISK-27345 - res_pjsip_session: RTP instances leak on 488
-       responses.
-       (Reported by Corey Farrell)
-  * ASTERISK-27337 - chan_sip: Security vulnerability with client
-       code header (revisited)
-       (Reported by Richard Mudgett)
-  * ASTERISK-27319 - (Security) Function in PJSIP 2.7
-       miscalculates the length of an unsigned long variable in 64bit
-       machines
-       (Reported by Kim youngsung)
-  * ASTERISK-27391 - Regression: Deadlock between AOR named lock
-       and pjproject grp lock
-       (Reported by shaurya jain)
-  * ASTERISK-27393 - res_pjsip: Crash occurs when an empty
-       contact read from astdb or database
-       (Reported by Aaron An)
-  * ASTERISK-27290 - res_pjsip: PIDF contact field has
-       malformed/invalid XML
-       (Reported by basildane)
-  * ASTERISK-27032 - res_pjsip: TLS options do not handle empty
-       values
-       (Reported by seanchann.zhou)
-  * ASTERISK-27395 - srtp: Add support for ephemeral DTLS
-       certificates
-       (Reported by Sean Bright)
-  * ASTERISK-26426 - format_ogg_opus: remove from source
-      
-       (Reported by Kevin Harwell)
-  * ASTERISK-27394 - [patch] tcptls: Print notice when TLS is
-       enabled but not configured.
-       (Reported by Alexander Traud)
-  * ASTERISK-27356 - [patch] libsrtp-2.x.x + AES-GCM support
-     
-       (Reported by Alexander Traud)
-  * ASTERISK-27378 - Modules: Fix issues with CLI completion.
-    
-       (Reported by Corey Farrell)
-  * ASTERISK-27387 - Regression: pjsip 13.18.0 - from_user - "+"
-       character isn't allowed any more
-       (Reported by Michael
-       Maier)
-  * ASTERISK-27364 - channel: Crash when fax gateway is in use
-       with PJSIP
-       (Reported by Jared Hull)
-  * ASTERISK-27390 - Audit menuselect module dependencies
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27389 - Optional API modules should not allow
-       unload.
-       (Reported by Corey Farrell)
-  * ASTERISK-27369 - Bridge() dialplan application fails without
-       setting BRIDGERESULT channel variable
-       (Reported by James
-       Terhune)
-  * ASTERISK-27067 - res_ari_channels: channel_state_invalid
-       always leaks snapshot reference.
-       (Reported by Marin
-       Odrljin)
-  * ASTERISK-27379 - stream: Allow streams on a topology to be
-       put into groups
-       (Reported by Joshua Colp)
-  * ASTERISK-27374 - alembic: PJSIP scripts are missing column
-       bundle in ps_endpoints table
-       (Reported by Florian
-       Floimair)
-  * ASTERISK-27377 - Typo in CHANNEL(dtmf_features) usage
-       documentation
-       (Reported by Igor Goncharovsky)
-  * ASTERISK-27181 - GCC 7 warning: app_voicemail.c: In function
-       'imap_delete_old_greeting'
-       (Reported by Anthony Messina)
-  * ASTERISK-27194 - jitterbuffer: Does not handle case where
-       translator returns null frame.
-       (Reported by Joshua Elson)
-  * ASTERISK-27372 - ARI: Node ARI client broken in latest
-       versions of 13 and 14
-       (Reported by Benjamin Keith Ford)
-  * ASTERISK-26639 - core: Disabling xmldoc support does not
-       work. Also results in abort during Asterisk startup.
-      
-       (Reported by Mr Dini)
-  * ASTERISK-18140 - Expires handling in SUBSCRIBE confuses the
-       absence of the Expires header field with an unsubscribe action.
-
-       (Reported by Jonathan Cloots)
-  * ASTERISK-25960 - The config_hook unit test causes Asterisk to
-       crash if run a second time
-       (Reported by George Joseph)
-  * ASTERISK-27198 - res_pjsip: SDP contains IP4 instead of IP6
-       when rtp_ipv6 set to yes
-       (Reported by Martin Cisárik)
-  * ASTERISK-27346 - res_xmpp: Crash if OAuth 2.0 is used before
-       curl is loaded
-       (Reported by Ronald Raikes)
-  * ASTERISK-27365 - [patch] chan_sip: Crypto attribute not last
-       but first on SDP media level.
-       (Reported by Alexander
-       Traud)
-  * ASTERISK-24483 - res_pjsip_pubsub.so, res_pjsip_refer.so:
-       Assertion on un/re-load: mod.id == -1
-       (Reported by Tzafrir
-       Cohen)
-  * ASTERISK-23462 - Cannot disable SIP debugging via CLI after
-       enabling with conf file option - also 'sip set debug off'
-       reports debugging disabled, when it really isn't
-       (Reported
-       by Rusty Newton)
-  * ASTERISK-27354 - bridge_softmix: When a channel leaves add in
-       any missing participant streams
-       (Reported by Joshua Colp)
-  * ASTERISK-27333 - sip_to_pjsip not correctly handling
-       disallow=all directive
-       (Reported by Torrey Searle)
-  * ASTERISK-27328 - Missing openssl dependencies in
-       res_rtp_asterisk and tcptls
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-27343 - Fails to build in FreeBSD due to
-       sys/sysmacros.h not existing there
-       (Reported by Guido
-       Falsi)
-  * ASTERISK-27341 - [patch] res_pjsip_session: SIP/SDP origin
-       (o=) contains local address.
-       (Reported by Alexander Traud)
-  * ASTERISK-27259 - chan_pjsip: Outgoing leg does not use all
-       configured codecs, but subset based on caller
-       (Reported by
-       lvl)
-  * ASTERISK-27340 - backtrace.c: Crash due to double-free.
-      
-       (Reported by Corey Farrell)
-  * ASTERISK-27339 - [patch] Crash on ast_ssl_teardown when
-       stopping.
-       (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-  * ASTERISK-24297 - cdr.c: Minor code optimizations.
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27449 - [PATCH] When failing to acquire target
-       during attended transfer, display wanted extension
-      
-       (Reported by Niklas Larsson)
-  * ASTERISK-27456 - app_voicemail: Add new object for
-       VoicemailUserEntry
-       (Reported by sungtae kim)
-  * ASTERISK-27380 - ast_coredumper: allow pointing out the
-       asterisk binary explicitly
-       (Reported by Tzafrir Cohen)
-  * ASTERISK-23556 - Compilation warning for invert.c (array
-       subscript is above array bounds)
-       (Reported by Marcello
-       Ceschia)
-  * ASTERISK-27355 - Upgrade bundled PJPROJECT to 2.7
-      
-       (Reported by Richard Mudgett)
-  * ASTERISK-27335 - CDR performance needs improvement.
-      
-       (Reported by Richard Mudgett)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="644fd46c3778650af8769ad97913bc4d0173dbebdbb538f6806f092a03366b2b" name="asterisk" arch="src">
  <version epoch="0" ver="15.6.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="313778580d225e1a5f2c281329c448c4ae9d2e3ec054d53e3a966c1d15dda40a" name="asterisk" arch="src">
  <version epoch="0" ver="15.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
</package>
<package pkgid="08b5c1b04223e733e5732b64285435008d56693c9d13310c512e3844f4c9c20c" name="asterisk" arch="src">
  <version epoch="0" ver="15.7.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9a2cb25d84f3b19019a56d9f6e01cc11857fcbdc22050dee7537a1a75fb0d66d" name="asterisk" arch="src">
  <version epoch="0" ver="15.7.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.2-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="bd37344004bcfa1944578a184da31545ab232c098f9d1c37b9d4971655042f9f" name="asterisk" arch="src">
  <version epoch="0" ver="15.7.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.2-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.3-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
- 
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3809b91a8466d369323bb67df4242cbfd15287948fee0abc79242c97320faa87" name="asterisk" arch="src">
  <version epoch="0" ver="15.7.4" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.2-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 15.7.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported
-                    by Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.0
-
- Thank you for your continued support of Asterisk!
- Asterisk Development Team &lt;asteriskteam@digium.com&gt;</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 15.7.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.2-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.3-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
- 
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.7.4-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions 13.28.1,
- 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-004: Crash when negotiating for T.38 with a declined stream
- When Asterisk sends a re-invite initiating T.38 faxing, and the endpoint
- responds with a declined media stream a crash will then occur in Asterisk.
-
- AST-2019-005: Remote Crash Vulnerability in audio transcoding
- When audio frames are given to the audio transcoding support in Asterisk the
- number of samples are examined and as part of this a message is output to
- indicate that no samples are present. A change was done to suppress this
- message for a particular scenario in which the message was not relevant. This
- change assumed that information about the origin of a frame will always exist
- when in reality it may not.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c536131edcc34e5e94f8266f4b24622701eed13ef02dc5742af9bb4fcd47025a" name="asterisk" arch="src">
  <version epoch="0" ver="16.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-1" date="1518523200">- The Asterisk Development Team would like to announce the release of Asterisk 15.2.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.2.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="699da3c184b8cb5f92b3e84d03323bb9d9063ff40a8b7b4feed1a831d9077f9b" name="asterisk" arch="src">
  <version epoch="0" ver="16.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.1-2" date="1518782400">- Added OPUS codec with transcoding support
- Added VP8 codec with passthrough support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b552b7f9e55c5d50420b18211e7380cfc232ba8e5ba914a1df51c3c76bdf02f1" name="asterisk" arch="src">
  <version epoch="0" ver="16.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.2.2-1" date="1519214400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.18. The available releases are
- released as versions 13.19.2, 14.7.6, 15.2.2 and 13.18-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2018-001: Crash when receiving unnegotiated dynamic payload
-     The RTP support in Asterisk maintains its own registry of dynamic codecs and
-     desired payload numbers. While an SDP negotiation may result in a codec using
-     a different payload number these desired ones are still stored internally.
-     When an RTP packet was received this registry would be consulted if the
-     payload number was not found in the negotiated SDP. This registry was
-     incorrectly consulted for all packets, even those which are dynamic. If the
-     payload number resulted in a codec of a different type than the RTP stream
-     (for example the payload number resulted in a video codec but the stream
-     carried audio) a crash could occur if no stream of that type had been
-     negotiated. This was due to the code incorrectly assuming that a stream of the
-     type would always exist.
-
-     AST-2018-002: Crash when given an invalid SDP media format description
-     By crafting an SDP message with an invalid media format description Asterisk
-     crashes when using the pjsip channel driver because pjproject's sdp parsing
-     algorithm fails to catch the invalid media format description.
-
-     AST-2018-003: Crash with an invalid SDP fmtp attribute
-     By crafting an SDP message body with an invalid fmtp attribute Asterisk
-     crashes when using the pjsip channel driver because pjproject's fmtp retrieval
-     function fails to check if fmtp value is empty (set empty if previously parsed
-     as invalid).
-
-     AST-2018-004: Crash when receiving SUBSCRIBE request
-     When processing a SUBSCRIBE request the res_pjsip_pubsub module stores the
-     accepted formats present in the Accept headers of the request. This code did
-     not limit the number of headers it processed despite having a fixed limit of
-     32. If more than 32 Accept headers were present the code would write outside
-     of its memory and cause a crash.
-
-     AST-2018-005: Crash when large numbers of TCP connections are closed suddenly
-     A crash occurs when a number of authenticated INVITE messages are sent over
-     TCP or TLS and then the connection is suddenly closed. This issue leads to a
-     segmentation fault.
-
-     AST-2018-006: WebSocket frames with 0 sized payload causes DoS
-     When reading a websocket, the length was not being checked. If a payload of
-     length 0 was read, it would result in a busy loop that waited for the
-     underlying connection to close.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.19.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.6
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert3
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="326ff63afb56dcafd1f0019e4c43adba16607774e5d9a2c5f5a94e75c85b854b" name="asterisk" arch="src">
  <version epoch="0" ver="16.1.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.3.0-1" date="1522152000">- The Asterisk Development Team would like to announce the release of Asterisk 15.3.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS
-                    (Reported by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not installed.
-                    (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by stasis_channel.c
-                    (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event (Reported
-                    by sungtae kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua Colp)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set but
-                    not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header mysql_version.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place 
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in version
-                    5.3. (Reported by Alexander Traud)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against header
-                    if match by ip fails (Reported by George Joseph)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes value
-                    to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an uninitialized
-                    variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and zero
-                    character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive values
-                    as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after unref
-                    (Reported by Richard Kenner)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch]contrib/scripts: add a way to migrate from chan_sip
-                    to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported by
-                    Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko Kamata)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5a8f10f439ff1f7ccec7262f7ddb6628439bdf341ef2a20825c2945ffc0c3f7b" name="asterisk" arch="src">
  <version epoch="0" ver="16.10.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="93ab7a42927393822afc882b19d16b479e3f10947a9ee74efc6dd75fc62d67ce" name="asterisk" arch="src">
  <version epoch="0" ver="16.11.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b7671ca381f73d168f7e68fa85cff3f468c628b136df8aab45e7d245d0e8d941" name="asterisk" arch="src">
  <version epoch="0" ver="16.11.1" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f60e955379db45c5ec2e4514cb0ed7ab343ab98ee08601f8d3cafc9e0413fc13" name="asterisk" arch="src">
  <version epoch="0" ver="16.12.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4485cd88895527eece3c6ab9b34f04d2e9eced7d12ef5497df18a6a55cc68255" name="asterisk" arch="src">
  <version epoch="0" ver="16.13.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8f26357aa6c0e60496adcfa8faee3d7af25ec643419e4f6b856fd67ab7dd0e40" name="asterisk" arch="src">
  <version epoch="0" ver="16.14.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="22c53c6ba74af2dd75d4a2d69d773294c88142e728a48b4252982674c3f55597" name="asterisk" arch="src">
  <version epoch="0" ver="16.14.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d6183d9fcec295198a17e77c85f9b572ba21ed7cba240686cd67cd15e38b1aa9" name="asterisk" arch="src">
  <version epoch="0" ver="16.15.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ba90b412ac0936fa5b653623bbca72d7552dd15de5cbf380eb0b7315d02cc552" name="asterisk" arch="src">
  <version epoch="0" ver="16.15.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f4beeaa96c38435b023647dd8c6a5db9850851cfc8b51e5e9870ea7923c7cd4d" name="asterisk" arch="src">
  <version epoch="0" ver="16.16.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.10.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="49021ebc9e4a3ce2b8437758088f5bf43a78cf859bb1b4e541e68ff92d1814bc" name="asterisk" arch="src">
  <version epoch="0" ver="16.16.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="08b47d0a49cbb7607e27c474551a524227c9b7274b1ea7074467e0b893318e2a" name="asterisk" arch="src">
  <version epoch="0" ver="16.16.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.11.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.11.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.11.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5f3875054bf19f430554ae9f0c26dde3ad8e02718f8d64a574246470d32ff395" name="asterisk" arch="src">
  <version epoch="0" ver="16.17.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.12.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.12.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b0562da4a3dfb8cccd10835a766a1c35e95c597dc99902a0ac12ddde13a1df75" name="asterisk" arch="src">
  <version epoch="0" ver="16.18.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.13.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.13.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="266ee0a75477973a1ccaa27a0d8c7c896dd6bded202f7970812937679d7489c2" name="asterisk" arch="src">
  <version epoch="0" ver="16.19.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.14.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29029] - Voicemail "pollmailboxes"-option not working, bug in
-                    function handle_subscribe (Reported by Karsten Wemheuer)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c8d39527c101e20b2e733e3ec90c0dbccf89fb7230bfba327030fbe8838df87e" name="asterisk" arch="src">
  <version epoch="0" ver="16.19.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.14.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="714dbb5eeb771036aa2c0fcd01f21062b0269959ccf84ec3aa2335fdb1a02704" name="asterisk" arch="src">
  <version epoch="0" ver="16.19.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 16.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="390ed090be333077cb7b0c2375fc10f4d8c313abc10b1258b412a23d1ff29625" name="asterisk" arch="src">
  <version epoch="0" ver="16.19.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.15.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
</package>
<package pkgid="c55a8d5b937c5f62967831e2dbaff81e94ddbf6b2e3b3c6d5297a5b0029d4b07" name="asterisk" arch="src">
  <version epoch="0" ver="16.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.0-1" date="1525176000">- The Asterisk Development Team would like to announce the release of Asterisk 15.4.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- -----------------------------------
-
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf
-                    (Reported by Richard Mudgett)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved address
-                    (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect (Reported
-                    by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not from
-                    glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not require
-                    libs for internal editline. (Reported by Alexander Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported by
-                    Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by Ross
-                    Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries. (Reported
-                    by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in ./configure.
-                    (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled with
-                    LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported by
-                    Jean Aunis - Prescom)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported by
-                    Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ec4194eddce60db97ed3285ec413167d9f45bfa3a26b874a6c97eca13d8c154d" name="asterisk" arch="src">
  <version epoch="0" ver="16.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.4.1-1" date="1529150400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15, 13 and 14, and Certified Asterisk 13.18 and 13.21. The available
- releases are released as versions 15.4.1, 13.21.1, 14.7.7, 13.18-cert4 and
- 13.21-cert2.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2018-007: Infinite loop when reading iostreams
-   When connected to Asterisk via TCP/TLS if the client abruptly disconnects, or
-   sends a specially crafted message then Asterisk gets caught in an infinite
-   loop while trying to read the data stream. Thus rendering the system as
-   unusable.
-
- AST-2018-008: PJSIP endpoint presence disclosure when using ACL
-   When endpoint specific ACL rules block a SIP request they respond with a 403
-   forbidden. However, if an endpoint is not identified then a 401 unauthorized
-   response is sent. This vulnerability just discloses which requests hit a
-   defined endpoint. The ACL rules cannot be bypassed to gain access to the
-   disclosed endpoints.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.4.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.21.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.7
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.18-cert4
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert2
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2018-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b29292a96ca7dc29f008af4c06a9a5ca6c389b5b00ea7a3c1d9dcbf80beb68ae" name="asterisk" arch="src">
  <version epoch="0" ver="16.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.16.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="066fb420e69de51172029211a9f52f47a8bc6021b0b4dc0541dab98537c8b9b6" name="asterisk" arch="src">
  <version epoch="0" ver="16.20.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.10.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
</package>
<package pkgid="8947eb40468e487b0d9187f37426d56ff3cd0b20a1d9651f03c75724956153b8" name="asterisk" arch="src">
  <version epoch="0" ver="16.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.16.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="97126f12d718018f9b2ed99da773bc91936e7583b0ad6b8f91ca9aa520450adc" name="asterisk" arch="src">
  <version epoch="0" ver="16.21.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.17.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video).
-                    (Reported by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again.  (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP.  (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang.  (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4ecd6f21c228e08fd32163bbcad8a68721de0be531ce85a0fb4786c4a47f2dbc" name="asterisk" arch="src">
  <version epoch="0" ver="16.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.18.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.18.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.18.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.18.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7d660991d22ac216fb5a6e362da6c01a09730e1beaced7c9a8073e294f37abfc" name="asterisk" arch="src">
  <version epoch="0" ver="16.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.19.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.19.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.19.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
</package>
<package pkgid="f30c5814cf069fdafcdf02d38126f6d55e63c70c6e46db95704b49291d538a18" name="asterisk" arch="src">
  <version epoch="0" ver="16.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="dc32364099dda9ad667e73de70bd6001c02b7fc956d81b29af048fe715f46993" name="asterisk" arch="src">
  <version epoch="0" ver="16.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="bf7d835a67736c9bdaefc803444cc7135fa8d20a9e0a0195c500712772165220" name="asterisk" arch="src">
  <version epoch="0" ver="16.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.19.1-3" date="1627992000">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="084a92f44539c5690374f02985e2442eab3901084864e740964e800174cc195c" name="asterisk" arch="src">
  <version epoch="0" ver="16.25.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.20.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.20.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.20.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fc8f688ed545ea132e9e68bb2cd17339c40f078cd0fa0b38aba22e8a97a364d5" name="asterisk" arch="src">
  <version epoch="0" ver="16.25.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.20.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a32f673c296cdf8916f6c8b4244dd12c9687b05c6b93d9852d459ac796a43d48" name="asterisk" arch="src">
  <version epoch="0" ver="16.25.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a87bebabb92caf56c56354f99388552f98b86b66af5d5b82a1bb16567842ba45" name="asterisk" arch="src">
  <version epoch="0" ver="16.26.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.21.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.21.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.21.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.21.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b8570628edeb20f72f8c4308d30ddf599bd3b4699752a21602bacc9ba46f4fe8" name="asterisk" arch="src">
  <version epoch="0" ver="16.26.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.22.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.22.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.22.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.22.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="015369c9b6b56d9c74e87c32489f8142dda67b22f3f1cca9b5a790c8fcfd698b" name="asterisk" arch="src">
  <version epoch="0" ver="16.27.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.23.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.23.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.23.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.23.0
-
- Thank you for your continued support of Asterisk!
-</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fa9e0460033f5297ce1eb66fa01c680be44298c23d58fc97075fa047cd6480c6" name="asterisk" arch="src">
  <version epoch="0" ver="16.28.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.24.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.24.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.24.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="62e1c5d099c735edb78852bd44c937bc022180474b6f0dbabc754d512e36231a" name="asterisk" arch="src">
  <version epoch="0" ver="16.28.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.24.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="e3a6fbe5035bc8ca9d9a74ce7a321cecaaa9685062766bf276cfe90ab63358b2" name="asterisk" arch="src">
  <version epoch="0" ver="16.28.0" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="626936a50c05143c6e32ce944ffaa72821a74486be88b8710c42d66146d67383" name="asterisk" arch="src">
  <version epoch="0" ver="16.29.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="afa939486e9df3276a9745b52a365355475560ab89d97837e5205d53cbcd1d91" name="asterisk" arch="src">
  <version epoch="0" ver="16.29.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="01388162fc01b1ef156c39d287c8d0c7c3fa3b344c8acc8bd992cd99678763a6" name="asterisk" arch="src">
  <version epoch="0" ver="16.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.5.0-1" date="1531483200">- The Asterisk Development Team would like to announce the release of Asterisk 15.5.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels are
-                    not removed if conference is ended as soon as it starts
-                    (Reported by Robert Mordec)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George Joseph)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported by
-                    Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before terminating
-                    nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax gateway
-                    (Reported by David Brillert)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27860] - [patch] res_pjsip: Register pjsip_transport_management not
-                    externally but internally. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as being
-                    minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by Florian
-                    Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is enabled
-                    only on the module (Reported by Marco Giordani)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua Colp)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList AMI
-                    response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided hostname
-                    (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="623ef035cbaea080273a5705d8a4b685d37c164d18eb6b00fe707f154c52200f" name="asterisk" arch="src">
  <version epoch="0" ver="16.30.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.25.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.25.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.25.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.25.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.30.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.30.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.30.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="78f7ba4af0f872ab89315f009b54e26f42d156d4e684f265e324d933ebe9e8ad" name="asterisk" arch="src">
  <version epoch="0" ver="16.30.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.26.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.26.1-1" date="1652961600">- sterisk Development Team would like to announce the release of Asterisk
- 16.26.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.26.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.27.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.27.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.27.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.27.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.28.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.28.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.28.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.28.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.29.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.29.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.30.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.30.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.30.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.30.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 16.30.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/16.30.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 16.30.1
- ==============================
-
- Links:
- ------
-
-  [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.30.1.md)
-  [GitHub Diff](https://github.com/asterisk/asterisk/compare/16.30.0...16.30.1)
-  [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-16.30.1.tar.gz)
-  [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
-  apply_patches: Use globbing instead of file/sort.
-  bundled_pjproject: Backport 2 SSL patches from upstream
-  bundled_pjproject: Backport security fixes from pjproject 2.13.1
-  apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list
-              before applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="33b4384d70c7b98d2717637849e6633c61d1f1aceb1fc10511c335d2a4521067" name="asterisk" arch="src">
  <version epoch="0" ver="16.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.0-1" date="1536148800">- The Asterisk Development Team would like to announce the release of Asterisk 15.6.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 15.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- -----------------------------------
-
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28011] - chan_sip: get_refer_info() attempted unlock mutex 'peer'
-                    without owning it! (Reported by Alec Davis)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is set
-                    (Reported by George Joseph)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in offer
-                    (Reported by Torrey Searle)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results in
-                    488 instead of declining stream (Reported by Thiago Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58 (Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management (Reported
-                    by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris 11.
-                    (Reported by Alexander Traud)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="15d35c0dd1ce3b9e7edd1dc26fc6271215f1a6564a67d16bc80aed8f61828d7d" name="asterisk" arch="src">
  <version epoch="0" ver="16.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 15.6.1-1" date="1537444800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 14 and 15, and Certified Asterisk 13.21. The available releases are
- released as versions 13.23.1, 14.7.8, 15.6.1 and 13.21-cert3.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- [AST-2018-009]: Remote crash vulnerability in HTTP websocket upgrade. there is
-                 a stack overflow vulnerability in the res_http_websocket.so
-                 module of Asterisk that allows an attacker to crash Asterisk
-                 via a specially crafted HTTP request to upgrade the connection
-                 to a websocket. The attacker’s request causes Asterisk to run
-                 out of stack space and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.23.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-14.7.8
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert3
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="05950a3cf3032a848c25293e086ce439d2483c1b02b029c7c39b91949c958c3c" name="asterisk" arch="src">
  <version epoch="0" ver="16.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.0-1" date="1539086400">- The Asterisk Development Team would like to announce the release of Asterisk 16.0.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade
-                    requests (Reported by Sean Bright)
- [ASTERISK-27807] - iostreams: Potential DoS when client connection closed
-                    prematurely (Reported by Sean Bright)
- [ASTERISK-27818] - Username bruteforce is possible when using ACL with PJSIP
-                    (Reported by John)
- [ASTERISK-27658] - WebSocket frames with 0 sized payload causes DoS (Reported
-                    by Sean Bright)
- [ASTERISK-27583] - Segmentation fault occurs in asterisk with an invalid SDP
-                    fmtp attribute (Reported by Sandro Gauci)
- [ASTERISK-27582] - Segmentation fault occurs in Asterisk with an invalid SDP
-                    media format description (Reported by Sandro Gauci)
- [ASTERISK-27618] - Crash occurs when sending a repeated number of INVITE
-                    messages over TCP or TLS transport (Reported by Sandro
-                    Gauci)
- [ASTERISK-27640] - SUBSCRIBE message with a large Accept value causes stack
-                    corruption (Reported by Sandro Gauci)
-
- New Features made in this release:
- -----------------------------------
- [ASTERISK-27286] - Add the ability to read the media file type from HTTP
-                    header for playback (Reported by Gaurav Khurana)
- [ASTERISK-27704] - Add cache_pools debug option to pjproject.conf (Reported
-                    by Richard Mudgett)
- [ASTERISK-27581] - Add new AMI Action for PJSIPShowContacts (Reported by
-                    Sungtae Kim)
- [ASTERISK-27547] - res_pjsip: Add new AMI Action for PJSIPShowAuths (Reported
-                    by Sungtae Kim)
- [ASTERISK-27117] - core: Add support for timelen parsing to ast_parse_arg and
-                    ACO. (Reported by Corey Farrell)
- [ASTERISK-27478] - PJSIP: Add CHANNEL(pjsip,request_uri) to get incoming
-                    INVITE Request-URI. (Reported by Richard Mudgett)
- [ASTERISK-27413] - Add cache_media_frames debugging option. (Reported by
-                    Richard Mudgett)
- [ASTERISK-27206] - res_pjsip: No mechanism exists to limit endpoint
-                    identification to IP only (Reported by Ben Merrills)
- [ASTERISK-27215] - [patch] AMI: Add CancelAtxfer Action (Reported by Thomas
-                    Sevestre)
- [ASTERISK-27322] - [New Feature] Add mute and DTMF passthrough to ARI add
-                    channel to bridge (Reported by Darren Sessions)
- [ASTERISK-27162] - [patch] chan_sip: Access incoming SIP REFER headers in the
-                    dialplan (Reported by Kirill Katsnelson)
- [ASTERISK-27163] - chan_sip: Dialplan function SIP_HEADERS() to complement
-                    SIP_HEADER(). (Reported by Kirill Katsnelson)
- [ASTERISK-27063] - Add support for systemd socket activation (Reported by
-                    Corey Farrell)
-
- Bugs fixed in this release:
- -----------------------------------
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported
-                    by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field
-                    is integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27880] - [patch] pjproject_bundled: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27810] - BASIC-RETRANS: Implement receive (Reported by Benjamin
-                    Keith Ford)
- [ASTERISK-27972] - res_sorcery_config: Allow object name based matching
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27965] - module: Remove old modules, update support levels
-                    (Reported by Joshua C. Colp)
- [ASTERISK-25548] - stasis: Improve message type "Use of before init/after
-                    destruction" error (Reported by Joshua C. Colp)
- [ASTERISK-27967] - srtp: rejecting short sdes lifetimes incompatible with
-                    obihai ATAs (Reported by Nick French)
- [ASTERISK-27961] - res_pjsip: Spurious ERROR logging when printing headers in
-                    sip_msg (Reported by Nick French)
- [ASTERISK-27563] - pjsip modules always get -O2 even when DONT_OPTIMIZE is
-                    set (Reported by George Joseph)
- [ASTERISK-27347] - [patch] pjproject_bundled: Disable TCP/TLS keep-alives.
-                    (Reported by Alexander Traud)
- [ASTERISK-27957] - PJSIP proposes ICE candidates on answer even if not in
-                    offer (Reported by Torrey Searle)
- [ASTERISK-27938] - [patch] Compile fails with `IPTOS_MINCOST' undeclared.
-                    (Reported by Alexander Traud)
- [ASTERISK-27955] - res_pjsip_session: sdp group:BUNDLE attribute truncated
-                    (Reported by Kevin Harwell)
- [ASTERISK-27956] - res_pjsip_pubsub: segfault in function publish_expire
-                    (Reported by Alexei Gradinari)
- [ASTERISK-27949] - res_pjsip_rfc3326: A lot of endpoints do not correctly
-                    handle two Reason headers (Reported by Ross Beer)
- [ASTERISK-27763] - res_pjsip_session: Initial INVITE with audio+fax results
-                    in 488 instead of declining stream (Reported by Thiago
-                    Coutinho)
- [ASTERISK-27657] - res_pjsip_t38: ATA fails with hangupcause 58(Bearer
-                    capability not available) (Reported by Jared Hull)
- [ASTERISK-27080] - res_pjsip_t38: Slow T.38 re-invite rejection if remote leg
-                    has T.38 disabled (Reported by Torrey Searle)
- [ASTERISK-26686] - res_pjsip: Lock inversion in transport management
-                    (Reported by Ross Beer)
- [ASTERISK-27939] - [patch] bridge_softmix_binaural: Enable FFTW3 in Solaris
-                    11. (Reported by Alexander Traud)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than
-                    100 before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-27783] - res_pjsip_pubsub: apparent crash on shutdown (Reported by
-                    Kevin Harwell)
- [ASTERISK-27870] - app_confbridge: Conference bridge and announcer channels
-                    are not removed if conference is ended as soon as it
-                    starts (Reported by Robert Mordec)
- [ASTERISK-27909] - cdr: Deadlock with submit_scheduled_batch and
-                    submit_unscheduled_batch (Reported by Denis Lebedev)
- [ASTERISK-26987] - pbx_dundi: Asterisk crashes when unloading module
-                    pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman)
- [ASTERISK-27943] - AMI: Action SendText needs to use the correct thread.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27942] - res_pjsip_messaging doesn't accept application/*
-                    content-types. (Reported by George Joseph)
- [ASTERISK-27936] - res_pjsip_session doesn't update media when a 200 comes in
-                    with a different port than a 183 (Reported by George
-                    Joseph)
- [ASTERISK-27933] - [patch] uuid: Enable UUID in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27625] - channels: CHECK_BLOCKING is ineffective (Reported by Corey
-                    Farrell)
- [ASTERISK-27931] - [patch] BuildSystem: Enable ./configure in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27926] - [patch] bootstrap.sh: find -maxdepth is not POSIX
-                    compatible. (Reported by Alexander Traud)
- [ASTERISK-27903] - menuselect: GCC 8: restrict-qualified parameter passed and
-                    aliased. (Reported by Alexander Traud)
- [ASTERISK-27914] - [patch] tests/test_utils: Repair ./configure
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27705] - chan_iax2: Stops listening for traffic (Reported by Kirsty
-                    Tyerman)
- [ASTERISK-27848] - rtp: DTMF Breaks With telephony-event/16000 (Reported by
-                    Dominic)
- [ASTERISK-27908] - [patch] crypto.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27905] - [patch] res_srtp: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27888] - SQL fetch error on query which return 0 columns (Reported
-                    by Alexei Gradinari)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-27901] - [patch] ooh323c: GCC 8: output truncated before
-                    terminating nul. (Reported by Alexander Traud)
- [ASTERISK-27872] - res_pjsip: Modified qualify_frequency doesn't effect until
-                    pjsip reload (Reported by Alexei Gradinari)
- [ASTERISK-27094] - res_fax: Deadlock when using Local channels and fax
-                    gateway (Reported by David Brillert)
- [ASTERISK-25261] - Manager events for MeetMe have incorrectly documented key
-                    name 'Usernum' - should be 'User' (Reported by Francois
-                    Blackburn)
- [ASTERISK-27878] - [patch] tcptls.h: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27876] - [patch] tcptls: Allow OpenSSL configured with no-dh.
-                    (Reported by Alexander Traud)
- [ASTERISK-27874] - [patch] tcptls: Allow OpenSSL 1.1.x configured with
-                    enable-ssl3-method no-deprecated. (Reported by Alexander
-                    Traud)
- [ASTERISK-27845] - Codec-Change Re-INVITE during DTMF can cause marker bit
-                    error (Reported by Torrey Searle)
- [ASTERISK-27831] - res_rtp_asterisk: Add support for abs-send-time RTP
-                    extension (Reported by Joshua C. Colp)
- [ASTERISK-27863] - config/ast_destroy_realtime_fields: successful DELETE is
-                    treated as failed (Reported by Alexei Gradinari)
- [ASTERISK-27865] - [patch]: tcptls: Repair ./configure --with-ssl=PATH.
-                    (Reported by Alexander Traud)
- [ASTERISK-27760] - Asterisk ODBC Voicemail Prompt storage fails with recent
-                    MariaDB version. (Reported by Nic Colledge)
- [ASTERISK-27853] - Incorrect error reported when leaving/retrieving a ODBC
-                    voicemail (Reported by Nic Colledge)
- [ASTERISK-27726] - chan_mobile: presents incorrect inbound Caller-ID names
-                    (Reported by Brian)
- [ASTERISK-27861] - [patch] res_pjsip_endpoint_identifier_ip: Unregister the
-                    module for headers. (Reported by Alexander Traud)
- [ASTERISK-27852] - cli: "manager show settings" mislabels HTTP timeout as
-                    being minutes. (Reported by Corey Farrell)
- [ASTERISK-27824] - Fix issues exposed by GCC 8 (Reported by George
-                    Joseph)
- [ASTERISK-27850] - [patch] rtp_engine: Allow Media Formats with
-                    add_static_payload(-1) on egress again. (Reported by
-                    Alexander Traud)
- [ASTERISK-27811] - [patch] sip_to_pjsip: Enable python3 compatibility.
-                    (Reported by Alexander Traud)
- [ASTERISK-27841] - digest over for manager (ami) over http fails on too long
-                    uris (Reported by Jaco Kroon)
- [ASTERISK-26570] - Macro allows an infinite loop of dialplan inclusion
-                    resulting in a crash (Reported by Tzafrir Cohen)
- [ASTERISK-27572] - cdr_mysql creates empty records if reconnects when mysql
-                    was not up on module load (Reported by Tzafrir Cohen)
- [ASTERISK-27801] - Asterisk got stuck while enabling "ari set debug all on"
-                    (Reported by shaurya jain)
- [ASTERISK-27795] - chan_sip: one way / no audio with srtp (Reported by
-                    Florian Kaiser)
- [ASTERISK-27800] - One way audio when calling from Asterisk(sip trunk) to
-                    another number where both are connected to a SBC using
-                    TLS+SRTP (Reported by Artur Pires)
- [ASTERISK-26806] - pjsip_options: rework to make more efficient (Reported by
-                    Kevin Harwell)
- [ASTERISK-27814] - translate: interpolated frames are not passed through
-                    (Reported by Kevin Harwell)
- [ASTERISK-27812] - When the ooh323 debug is on there is no ringing signal to
-                    incoming calls via H323 trunk. (Reported by Dimos)
- [ASTERISK-26893] - No "alert" or "progress" in chan_ooh323 if debug is
-                    enabled only on the module (Reported by Marco Giordani)
- [ASTERISK-27804] - bridge_softmix / app_confbridge: Add support for combining
-                    REMB reports (Reported by Joshua C. Colp)
- [ASTERISK-27639] - [patch] BuildSystem: Enable IMAP storage on FreeBSD and
-                    DragonFly BSD. (Reported by Alexander Traud)
- [ASTERISK-27418] - app_confbridge: "core show profile bridge" does not output
-                    "sfu" when video_mode is sfu (Reported by Carlos Chavez)
- [ASTERISK-27809] - [patch] utils/pval: Add -lBlocksRuntime for compiler clang
-                    conditionally. (Reported by Alexander Traud)
- [ASTERISK-27808] - [patch] chan_vpb: Avoid GNU old-style field designator
-                    extension. (Reported by Alexander Traud)
- [ASTERISK-27806] - BASIC-RETRANS: Implement send (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-27774] - res_musiconhold: Music on hold restarts after every
-                    announcement (Reported by lvl)
- [ASTERISK-27782] - cdr_mysql: Missing MYSQL_PORT definition (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27614] - res_pjsip_session: SDP origin does not use resolved
-                    address (Reported by John M.)
- [ASTERISK-27776] - res_rtp_asterisk: Add support for sending RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-27740] - chan_sip: New Channel creation from new SIP dialog with
-                    Replaces failed to be properly tracked and destroyed
-                    (Reported by Shannon Price)
- [ASTERISK-27786] - app_confbridge: Add ability to enable and configure REMB
-                    support (Reported by Joshua C. Colp)
- [ASTERISK-27706] - PJSIP: Deadlock shutting down subscription TCP connection
-                    and sending subscription message. (Reported by Ross Beer)
- [ASTERISK-27688] - res_pjsip: Crash on TCP PJSIP Transport Disconnect
-                    (Reported by Ross Beer)
- [ASTERISK-27758] - res_rtp_asterisk: Add support for raising RTCP feedback
-                    messages (Reported by Joshua C. Colp)
- [ASTERISK-26366] - rtp: RTCP messages with REMB trigger fast picture update
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27773] - Command line not being parsed correctly with getopt not
-                    from glibc (Reported by Guido Falsi)
- [ASTERISK-27435] - [patch] configure: pjsip_evsub_set_uas_timeout not found.
-                    (Reported by Alexander Traud)
- [ASTERISK-27761] - [patch] BuildSystem: With external editline, do not
-                    require libs for internal editline. (Reported by Alexander
-                    Traud)
- [ASTERISK-27755] - ConfBridge: raise ConfbridgeTalking when put on hold and
-                    clear talking status (Reported by Kevin Harwell)
- [ASTERISK-27743] - Generic PLC doesn't work if the 2 codecs on a channel are
-                    equal (Reported by George Joseph)
- [ASTERISK-27745] - [patch] BuildSystem: Remove unused dependency on libltdl.
-                    (Reported by Alexander Traud)
- [ASTERISK-12841] - [patch] Make format_ogg_vorbis work on OpenBSD (Reported
-                    by Michiel van Baak)
- [ASTERISK-27720] - [patch] BuildSystem: Enable Advanced Linux Sound
-                    Architecture (ALSA) in NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27741] - res_pjsip_rfc3326.c rfc3326_use_reason_header doesn't
-                    account for more than one 'Reason' header (Reported by
-                    Ross Beer)
- [ASTERISK-27734] - [patch] BuildSystem: Enable IMAP storage on openSUSE and
-                    Arch Linux. (Reported by Alexander Traud)
- [ASTERISK-27686] - [patch] install_prereq: Update FreeBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27733] - [patch] res_srtp: Add support for libsrtp2.x on openSUSE.
-                    (Reported by Alexander Traud)
- [ASTERISK-11015] - NetBSD Build Needs RPATH set in 1.2.25 (Reported by Curt
-                    Sampson)
- [ASTERISK-27641] - BuildSystem: Enable Better Backtraces in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27671] - Deprecate legacy modules (Reported by Corey
-                    Farrell)
- [ASTERISK-25586] - uuid_generate_random detection failure (Reported by John
-                    Nemeth)
- [ASTERISK-27721] - [patch] BuildSystem: Enable PortAudio in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27715] - [patch] BuildSystem: AC_PATH_PROG sets to colon character
-                    when not found. (Reported by Alexander Traud)
- [ASTERISK-27554] - res_pjsip_rfc3326: Order of 'Reason' headers break many
-                    endpoints (Reported by Ross Beer)
- [ASTERISK-27703] - AMI Action VoicemailUsersList returns 0 MessageCount
-                    (Reported by Sébastien Duthil)
- [ASTERISK-27674] - chan_sip: RTP framing issues on outgoing calls (Reported
-                    by Jean Aunis - Prescom)
- [ASTERISK-27441] - PJSIP: Forked INVITE SDP negotiation gets one way audio.
-                    (Reported by lvl)
- [ASTERISK-27718] - [patch] BuildSystem: Enable Lua in NetBSD. (Reported by
-                    Alexander Traud)
- [ASTERISK-27722] - [patch] BuildSystem: Depend not implicitly but explicitly
-                    on external libraries. (Reported by Alexander Traud)
- [ASTERISK-27719] - [patch] res_http_post: Enable GMime in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27716] - [patch] BuildSystem: Enable autotools in NetBSD. (Reported
-                    by Alexander Traud)
- [ASTERISK-27714] - [patch] chan_unistim: NetBSD has an incompatible struct
-                    in_pktinfo. (Reported by Alexander Traud)
- [ASTERISK-27713] - [patch] BuildSystem: Cast any intptr_t explicitly to its
-                    proposed type. (Reported by Alexander Traud)
- [ASTERISK-27712] - [patch] BuildSystem: Detect whether uselocale(.) is
-                    available. (Reported by Alexander Traud)
- [ASTERISK-27711] - [patch] BuildSystem: Avoid re-defining of pthread_* on
-                    NetBSD. (Reported by Alexander Traud)
- [ASTERISK-27710] - [patch] BuildSystem: Install init scripts on openSUSE
-                    Tumbleweed. (Reported by Alexander Traud)
- [ASTERISK-27709] - [patch] BuildSystem: Avoid == for comparison in
-                    ./configure. (Reported by Alexander Traud)
- [ASTERISK-27610] - app_amd.so returning TOOLONG before reaching the timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-26688] - Documentation: voicemail.conf.sample shows 512 limit for
-                    emailbody field, however this is only true if compiled
-                    with LOW_MEMORY option (Reported by Fran Vicente)
- [ASTERISK-27568] - PJSIP: Crash during SIP attended transfer. (Reported by
-                    Bryan Walters)
- [ASTERISK-27659] - Output from rawman truncated if output is long enough
-                    (Reported by Bojan Nemčić)
- [ASTERISK-27692] - bridging: Sometimes cloning the stream topology causes a
-                    crash (Reported by Richard Mudgett)
- [ASTERISK-27488] - core: If frame with unnegotiated format is read crash will
-                    occur (Reported by Sébastien Duthil)
- [ASTERISK-24488] - Wrong remote identity and target in dialog package XML in
-                    NOTIFY (Reported by Alejandro Padilla)
- [ASTERISK-24386] - Asterisk "doc/lang/language-criteria.txt" needs update or
-                    removal. (Reported by Rusty Newton)
- [ASTERISK-27646] - ICE fails with no candidate nominated (Reported by Thomas
-                    Guebels)
- [ASTERISK-27689] - [patch] rtp_engine: Load format name / mime type in
-                    uppercase again. (Reported by Alexander Traud)
- [ASTERISK-27679] - res_pjsip: Endpoint destruction does not free DTLS
-                    configuration (Reported by Mak Dee)
- [ASTERISK-27684] - [patch] install_prereq: Update OpenBSD libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27680] - [patch] res_calendar: Specialized calendars depend on
-                    symbols of general calendar. (Reported by Alexander Traud)
- [ASTERISK-27681] - [patch] BuildSystem: Enable IMAP storage on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27677] - [patch] BuildSystem: Enable system provided libedit on
-                    OpenBSD. (Reported by Alexander Traud)
- [ASTERISK-27670] - [patch] BuildSystem: Remove chan_h323 leftovers. (Reported
-                    by Alexander Traud)
- [ASTERISK-27595] - [patch] BuildSystem: Invoke ldconfig with previous paths.
-                    (Reported by Alexander Traud)
- [ASTERISK-27631] - [patch] BuildSystem: Do not warn when bash is not
-                    installed. (Reported by Alexander Traud)
- [ASTERISK-27666] - chan_sip: Crash processing CANCEL request (Reported by
-                    Leandro Dardini)
- [ASTERISK-27584] - Internal pjproject build doesn't disable bcg729 (Reported
-                    by Stuart Henderson)
- [ASTERISK-27669] - [patch] codecs: Add support for WebRTC iLBC 2.0. (Reported
-                    by Alexander Traud)
- [ASTERISK-27634] - Determine if the internal editline and stdtime libraries
-                    are still relevant (Reported by George Joseph)
- [ASTERISK-27642] - [patch] backtrace: Avoid -Wlogical-not-parentheses.
-                    (Reported by Alexander Traud)
- [ASTERISK-27555] - [patch] install_prereq: Update Debian/Ubuntu libraries.
-                    (Reported by Alexander Traud)
- [ASTERISK-27656] - CDR: Leaking channel snapshots allocated by
-                    stasis_channel.c (Reported by Kristijan Vrban)
- [ASTERISK-27426] - chan_console: cannot read and write at the same time with
-                    alsa backend (Reported by Tzafrir Cohen)
- [ASTERISK-27621] - (null) string tailing after AsyncAGIEnd AMI event
-                    (Reported by Sungtae Kim)
- [ASTERISK-27652] - Null pointer Crash in PJSIP MWI (Reported by Joshua
-                    Elson)
- [ASTERISK-27571] - res_pjsip: If SIP response is received during shutdown a
-                    crash may occur (Reported by Joshua C. Colp)
- [ASTERISK-27619] - Build System: Require compiler to provide built-in support
-                    for atomic references. (Reported by Corey Farrell)
- [ASTERISK-27612] - Subscriptions Persist After Expiration and TCP/TLS
-                    Disconnect (Reported by Ross Beer)
- [ASTERISK-27637] - [patch] BuildSystem: Enable autotools in FreeBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27635] - [patch] app_voicemail: Avoid always true warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-27599] - [patch] install_prereq: Update RHEL/CentOS/Fedora
-                    libraries. (Reported by Alexander Traud)
- [ASTERISK-26563] - core: macOS devmode build fails: variable 'freeswap' set
-                    but not used (Reported by David M. Lee)
- [ASTERISK-27630] - [patch] editline: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-16172] - Problems with siren14 codec; problems with siren7 sound
-                    files. (Reported by Steve Murphy)
- [ASTERISK-16951] - [patch] configure.ac in 1.4.37 broken with autoconf 2.60
-                    (Reported by Stéphan Kochen)
- [ASTERISK-27603] - [patch] install_prereq: Download latest Jansson. (Reported
-                    by Alexander Traud)
- [ASTERISK-27620] - New module loader aborts startup if a required module
-                    declines load. (Reported by snuffy)
- [ASTERISK-27607] - [patch] res_config_mysql: Avoid the header
-                    mysql_version.h. (Reported by Alexander Traud)
- [ASTERISK-24598] - When running ./contrib/scripts/install_prereq
-                    install-unpackaged pjproject is installed in wrong place
-                    (Reported by PowerPBX)
- [ASTERISK-27602] - [patch] BuildSystem: AC_CONFIG_AUX_DIR needs a directory.
-                    (Reported by Alexander Traud)
- [ASTERISK-27600] - [patch] BuildSystem: Allow make clean all again. (Reported
-                    by Alexander Traud)
- [ASTERISK-27598] - [patch] install_prereq: Support package manager DNF.
-                    (Reported by Alexander Traud)
- [ASTERISK-26596] - Placing call on hold temporarily locks up set (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-27596] - [patch] BuildSystem: Use the detected name for MD5
-                    everywhere. (Reported by Alexander Traud)
- [ASTERISK-27594] - [patch] BuildSystem: Invoke install not in GNU but POSIX
-                    style. (Reported by Alexander Traud)
- [ASTERISK-27593] - [patch] BuildSystem: In OpenBSD, xmlstarlet is xml.
-                    (Reported by Alexander Traud)
- [ASTERISK-27592] - [patch] BuildSystem: Detect external library Lua in
-                    version 5.3. (Reported by Alexander Traud)
- [ASTERISK-27491] - res_pjsip_endpoint_identifier_ip only matches against
-                    header if match by ip fails (Reported by George Joseph)
- [ASTERISK-26832] - res_pjsip: Segfault when calling pjsip_hdr_print_on in
-                    sip_msg.c:581 (Reported by Ross Beer)
- [ASTERISK-27589] - [patch] BuildSystem: Avoid $EUID and use id -u instead.
-                    (Reported by Alexander Traud)
- [ASTERISK-27585] - [patch] BuildSystem: Resolve resolv.h not via Generic but
-                    Particular Header-Check. (Reported by Alexander Traud)
- [ASTERISK-27575] - menuselect : remove obsolete TRACE_FRAMES compiler flag
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-27576] - [patch] res_config_pgsql: Avoid typecasting an int to
-                    unsigned char. (Reported by Alexander Traud)
- [ASTERISK-27560] - [patch] clang 5 does not know -Wno-format-truncation
-                    (Reported by Alexander Traud)
- [ASTERISK-27578] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-27577] - [patch] chan_ooh323: Avoid typecasting an int to unsigned
-                    short. (Reported by Alexander Traud)
- [ASTERISK-27534] - chan_sip: Assumes iostream is non-NULL when it may not be
-                    (Reported by Lubos Dolezel)
- [ASTERISK-27549] - [patch] translate: Avoid absolute value on unsigned
-                    substraction. (Reported by Alexander Traud)
- [ASTERISK-27566] - res_pjsip_session: Improve WebRTC interop with bundling
-                    during renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-27553] - [patch] res_curl: Avoid error message on unload. (Reported
-                    by Alexander Traud)
- [ASTERISK-27557] - [patch] clang 5.0: implicit conversion to char changes
-                    value to negative. (Reported by Alexander Traud)
- [ASTERISK-27550] - [patch] bridge_softmix: Avoid warning about an
-                    uninitialized variable. (Reported by Alexander Traud)
- [ASTERISK-27559] - [patch] editline: Avoid comparison between pointer and
-                    zero character constant. (Reported by Alexander Traud)
- [ASTERISK-27558] - [patch] codec_gsm: Avoid shifting a negative signed value.
-                    (Reported by Alexander Traud)
- [ASTERISK-25329] - Asterisk configure fails on 'cannot find ptlib-config',
-                    despite ptlib-config existing (Reported by Rusty Newton)
- [ASTERISK-27552] - [patch] chan_ooh323: Limit outgoinglimit to positive
-                    values as intended. (Reported by Alexander Traud)
- [ASTERISK-27551] - [patch] ooh323cDriver: Fix typo in header guard. (Reported
-                    by Alexander Traud)
- [ASTERISK-26046] - [patch] Avoid obsolete warnings on autoconf. (Reported by
-                    Alexander Traud)
- [ASTERISK-20346] - Modules need to ensure that any functions, apps, AMI
-                    actions, etc. they register are unregistered if the module
-                    declines loading (Reported by Mark Michelson)
- [ASTERISK-27539] - 'cdr submit' fails: batch mode not enabled. (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27498] - ICE candidate parser - ICE foundation parsing too short
-                    (Reported by Michele Prà)
- [ASTERISK-25128] - Datastore: Implement automatic module references.
-                    (Reported by Corey Farrell)
- [ASTERISK-27366] - Asterisk Turkish Language Set Problem (Reported by Halil
-                    İbrahim YILDIZ)
- [ASTERISK-23133] - Documentation fix - MASTER_CHANNEL Unexpected Behaviour
-                    (Reported by Shane Mitchell)
- [ASTERISK-27531] - Compiler optimizations can break module load sequence.
-                    (Reported by abelbeck)
- [ASTERISK-27480] - Security: Authenticated SUBSCRIBE without Contact crashes
-                    asterisk (Reported by Ross Beer)
- [ASTERISK-24198] - Typo's (Reported by Walter
-                    Doekes)
- [ASTERISK-27229] - bridge: Old channel video source not set to NULL after
-                    unref (Reported by Richard Kenner)
- [ASTERISK-27495] - DNS: Unexpected rr_type can cause crash (Reported by Corey
-                    Farrell)
- [ASTERISK-25079] - AMI bridge of channels results in MOH not destroyed and
-                    robotic audio on one channel (Reported by Zane Conkle)
- [ASTERISK-27490] - chan_console: 'set active' fails to work (Reported by
-                    Tzafrir Cohen)
- [ASTERISK-27299] - Asterisk Hangs with Bad file descriptor on read()
-                    (Reported by Abhay Gupta)
- [ASTERISK-24756] - ConfBridge sound_muted does not work from CLI or AMI
-                    (Reported by Thomas Frederiksen)
- [ASTERISK-25649] - Transfer application does not work with Local channels -
-                    documentation misleading (Reported by Ivan Ullmann)
- [ASTERISK-25869] - chan_sip: "rejected because extension not found" should be
-                    logged as a security event (Reported by Brian J. Murrell)
- [ASTERISK-27440] - Strictrtp has issues to qualify video rtp streams
-                    (Reported by Wim De Vlaminck)
- [ASTERISK-19657] - Coverity Report: Fix issues for error type CHAR_IO
-                    (Reported by Matt Jordan)
- [ASTERISK-27175] - iax.conf demo peer is invalid (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27430] - README refers to security documents that do not exist.
-                    (Reported by Corey Farrell)
- [ASTERISK-20281] - "core set verbose" behaves strangely, can't alias it,
-                    cli.conf example broken (Reported by Tim Ringenbach at
-                    Asteria Solutions Group)
- [ASTERISK-27382] - crash after an invalid rtcp packet from GT48 FXS gateway
-                    (Reported by Tzafrir Cohen)
- [ASTERISK-27429] - res_rtp_asterisk: Multiple reports in an RTCP packet will
-                    write past where it should (Reported by Vitezslav Novy)
- [ASTERISK-27408] - Identify causes and fix
-                    pjsip/resolver/srv/failover/in_dialog/transport_tcp
-                    (Reported by Corey Farrell)
- [ASTERISK-18411] - Queue members with hints for state_interface get stuck in
-                    "In Use" state. (Reported by Steven T. Wheeler)
- [ASTERISK-26131] - chan_sip: Crash Asterisk (in sip_request_call at
-                    chan_sip.c) by making a call to a single character in a
-                    dot pattern match (Reported by Dwayne Hubbard)
- [ASTERISK-27467] - pjsip_options: qualify_frequency sometimes not applied on
-                    reload (Reported by John Bigelow)
- [ASTERISK-27460] - CDR: Deadlock using AMI Originate with Variable
-                    CDR(amaflags)=... (Reported by Richard Mudgett)
- [ASTERISK-27453] - RTP: Blind transfer direct media scenario results in one
-                    way audio. (Reported by Richard Mudgett)
- [ASTERISK-20643] - SIP ICE support - remove hardcoded limitation on SDP size,
-                    make ICE support disabled by default in SIP, maybe provide
-                    a better warning message (Reported by Roy)
- [ASTERISK-27457] - chan_sip: Guests disallowed via TCP (or TLS) if existing
-                    peer from same IP. (Reported by Alexander Traud)
- [ASTERISK-26980] - pjsip: Clean up WebRTC disables (Reported by
-                    abelbeck)
- [ASTERISK-27452] - Security: chan_skinny: Memory exhaustion if flooded with
-                    unauthenticated requests (Reported by George Joseph)
- [ASTERISK-27454] - res_http_post: Don't require GMIME_MAJOR_VERSION (Reported
-                    by Joshua C. Colp)
- [ASTERISK-23735] - Transcoding makes bad choice in high-rate translations
-                    (Reported by Richard Kenner)
- [ASTERISK-27445] - ARI: Updating a bridge gives wrong error message.
-                    (Reported by Frank Durden)
- [ASTERISK-24662] - [patch] column and row headers for Signed Linear format
-                    variants in output of 'core show translation' are
-                    ambiguous (Reported by Rusty Newton)
- [ASTERISK-27353] - H323 audio starts with a delay of 2 seconds. (Reported by
-                    Marco Giordani)
- [ASTERISK-27442] - pjsip: 183 without To tag does not negotiate media
-                    (Reported by Kevin Harwell)
- [ASTERISK-27437] - [patch] ICE: server-reflexive candidates (srflx) with
-                    Dual-Stack. (Reported by Alexander Traud)
- [ASTERISK-27434] - [patch] chan_sip/ICE: Square brackets around IPv6
-                    addresses. (Reported by Alexander Traud)
- [ASTERISK-27332] - Asterisk fails to configure on MacOS Sierra (Reported by
-                    Ivan Larionov)
- [ASTERISK-27431] - Asterisk fails to build when openssl headers are not
-                    installed. (Reported by Corey Farrell)
- [ASTERISK-27421] - RTP source learning not working with devices that have
-                    some clock issues (Reported by nappsoft)
- [ASTERISK-27361] - Attended transfer crashes in Asterisk 13.17.2 (Reported by
-                    Alessandro Pimenta)
- [ASTERISK-27238] - Bridging: Crash freeing a frame that's already been freed
-                    (Reported by Richard Kenner)
- [ASTERISK-27412] - core: Audiohook freeing interpolated frame when it
-                    shouldn't. (Reported by Mikhail)
- [ASTERISK-27423] - app_record: We set the RECORD_STATUS channel variable
-                    before closing the file (Reported by George Joseph)
- [ASTERISK-26758] - res_hep_pjsip: For WebRTC clients Asterisk insert same ip
-                    address in "source ip address" and "destination ip
-                    address" fields in HEP packets (Reported by Max Norba)
- [ASTERISK-27363] - res_http_websocket: Wrong LocalAddress (it is equal to
-                    RemoteAddress) (Reported by Vasilii Rogin)
- [ASTERISK-27415] - asterisk.conf: Setting astctl without setting astrundir is
-                    ineffective. (Reported by Corey Farrell)
- [ASTERISK-27411] - pjsip: TCP connections may not be destroyed (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27404] - DEBUG_FD_LEAKS does not record socketpair, timerfd_create
-                    or eventfd. (Reported by Corey Farrell)
- [ASTERISK-27345] - res_pjsip_session: RTP instances leak on 488 responses.
-                    (Reported by Corey Farrell)
- [ASTERISK-27337] - chan_sip: Security vulnerability with client code header
-                    (revisited) (Reported by Richard Mudgett)
- [ASTERISK-27319] - (Security) Function in PJSIP 2.7 miscalculates the length
-                    of an unsigned long variable in 64bit machines (Reported
-                    by Kim youngsung)
- [ASTERISK-27391] - Regression: Deadlock between AOR named lock and pjproject
-                    grp lock (Reported by shaurya jain)
- [ASTERISK-27393] - res_pjsip: Crash occurs when an empty contact read from
-                    astdb or database (Reported by Aaron An)
- [ASTERISK-27290] - res_pjsip: PIDF contact field has malformed/invalid XML
-                    (Reported by basildane)
- [ASTERISK-27032] - res_pjsip: TLS options do not handle empty values
-                    (Reported by seanchann.zhou)
- [ASTERISK-27395] - srtp: Add support for ephemeral DTLS certificates
-                    (Reported by Sean Bright)
- [ASTERISK-26426] - format_ogg_opus: remove from source (Reported by Kevin
-                    Harwell)
- [ASTERISK-27394] - [patch] tcptls: Print notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-27356] - [patch] libsrtp-2.x.x + AES-GCM support (Reported by
-                    Alexander Traud)
- [ASTERISK-27378] - Modules: Fix issues with CLI completion. (Reported by
-                    Corey Farrell)
- [ASTERISK-27387] - Regression: pjsip 13.18.0 - from_user - "+" character
-                    isn't allowed any more (Reported by Michael Maier)
- [ASTERISK-27364] - channel: Crash when fax gateway is in use with PJSIP
-                    (Reported by Jared Hull)
- [ASTERISK-27390] - Audit menuselect module dependencies (Reported by Corey
-                    Farrell)
- [ASTERISK-27389] - Optional API modules should not allow unload. (Reported by
-                    Corey Farrell)
- [ASTERISK-27369] - Bridge() dialplan application fails without setting
-                    BRIDGERESULT channel variable (Reported by James Terhune)
- [ASTERISK-27067] - res_ari_channels: channel_state_invalid always leaks
-                    snapshot reference. (Reported by Marin Odrljin)
- [ASTERISK-27379] - stream: Allow streams on a topology to be put into groups
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27374] - alembic: PJSIP scripts are missing column bundle in
-                    ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27377] - Typo in CHANNEL(dtmf_features) usage documentation
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-27181] - GCC 7 warning: app_voicemail.c: In function
-                    'imap_delete_old_greeting' (Reported by Anthony Messina)
- [ASTERISK-27194] - jitterbuffer: Does not handle case where translator
-                    returns null frame. (Reported by Joshua Elson)
- [ASTERISK-27372] - ARI: Node ARI client broken in latest versions of 13 and
-                    14 (Reported by Benjamin Keith Ford)
- [ASTERISK-26639] - core: Disabling xmldoc support does not work. Also results
-                    in abort during Asterisk startup. (Reported by Mr Dini)
- [ASTERISK-18140] - Expires handling in SUBSCRIBE confuses the absence of the
-                    Expires header field with an unsubscribe action. (Reported
-                    by Jonathan Cloots)
- [ASTERISK-25960] - The config_hook unit test causes Asterisk to crash if run
-                    a second time (Reported by George Joseph)
- [ASTERISK-27198] - res_pjsip: SDP contains IP4 instead of IP6 when rtp_ipv6
-                    set to yes (Reported by Martin Cisárik)
- [ASTERISK-27346] - res_xmpp: Crash if OAuth 2.0 is used before curl is loaded
-                    (Reported by Ronald Raikes)
- [ASTERISK-27365] - [patch] chan_sip: Crypto attribute not last but first on
-                    SDP media level. (Reported by Alexander Traud)
- [ASTERISK-24483] - res_pjsip_pubsub.so, res_pjsip_refer.so: Assertion on
-                    un/re-load: mod.id == -1 (Reported by Tzafrir Cohen)
- [ASTERISK-23462] - Cannot disable SIP debugging via CLI after enabling with
-                    conf file option - also 'sip set debug off' reports
-                    debugging disabled, when it really isn't (Reported by
-                    Rusty Newton)
- [ASTERISK-27350] - app_macro deprecation (Reported by Corey
-                    Farrell)
- [ASTERISK-27354] - bridge_softmix: When a channel leaves add in any missing
-                    participant streams (Reported by Joshua C. Colp)
- [ASTERISK-27333] - sip_to_pjsip not correctly handling disallow=all directive
-                    (Reported by Torrey Searle)
- [ASTERISK-27343] - Fails to build in FreeBSD due to sys/sysmacros.h not
-                    existing there (Reported by Guido Falsi)
- [ASTERISK-27341] - [patch] res_pjsip_session: SIP/SDP origin (o=) contains
-                    local address. (Reported by Alexander Traud)
- [ASTERISK-27259] - chan_pjsip: Outgoing leg does not use all configured
-                    codecs, but subset based on caller (Reported by lvl)
- [ASTERISK-27340] - backtrace.c: Crash due to double-free. (Reported by Corey
-                    Farrell)
- [ASTERISK-27339] - [patch] Crash on ast_ssl_teardown when stopping. (Reported
-                    by Alexander Traud)
- [ASTERISK-27047] - res_pjsip: user=phone added to Anonymous caller-id when it
-                    shouldn't be. (Reported by dtryba)
- [ASTERISK-26988] - res_pjsip_session: user_eq_phone adds double user=phone
-                    parameters to URIs (Reported by dtryba)
- [ASTERISK-27301] - [patch] app_queue: Music On Hold for real-time queues is
-                    not reset to default (Reported by Nathan Bruning)
- [ASTERISK-25266] - Application Originate returns SUCCESS to ORIGINATE_STATUS
-                    upon failure to originate (Reported by Allen Ford)
- [ASTERISK-27270] - cdr_mysql: various crashes at second module reload if
-                    cdr_mysql.conf is configured (Reported by Tzafrir Cohen)
- [ASTERISK-27328] - Missing openssl dependencies in res_rtp_asterisk and
-                    tcptls (Reported by Tzafrir Cohen)
- [ASTERISK-27192] - res_pjsip: Loss of SIP registrations causing unavailable
-                    endpoints (Reported by Richard Mudgett)
- [ASTERISK-27305] - res_ari: Memory leaks in ARI when using Content-Type:
-                    application/json (Reported by David Hajek)
- [ASTERISK-26922] - chan_sip: tcpbind uses wrong source address (Reported by
-                    Ksenia)
- [ASTERISK-27324] - [patch] Dual-Stack server cannot be used as IPv4 client
-                    via TCP/TLS (Reported by Alexander Traud)
- [ASTERISK-27317] - vector: multiple evaluation of elem in
-                    AST_VECTOR_ADD_SORTED. (Reported by Corey Farrell)
- [ASTERISK-27318] - res_pjsip_mwi: uninitialized value from ast_strings_match
-                    (Reported by Corey Farrell)
- [ASTERISK-27284] - Status of RFC 3323 and PJSIP (Reported by
-                    dtryba)
- [ASTERISK-27296] - [patch] False positive busy checks when icalendar's
-                    recurrence-id mechanism is involved (Reported by Benoît
-                    Dereck-Tricot)
- [ASTERISK-27216] - app_queue: does its check-makeannouncement-logic twice
-                    each head-caller-loop (Reported by Stefan Engström)
- [ASTERISK-27298] - Problem with expires on pjsip / outbound-publish (Reported
-                    by Cyrille Demaret)
- [ASTERISK-27295] - Contact is improperly translated after d178f497 (Reported
-                    by Sean Bright)
- [ASTERISK-27292] - Multiple RTP Stream Created Breaking RFC2833 (SSRC
-                    Changes) (Reported by Ross Beer)
- [ASTERISK-27289] - A codeblock that maintains a bug,but maybe the codeblock
-                    will never run (Reported by Huangyx)
- [ASTERISK-27277] - bridge: Renegotiate if source stream changes. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27264] - res_pjsip_session: Crashes after sending PRACK and
-                    receiving 200 OK (Reported by Daniel Heckl)
- [ASTERISK-27283] - Realtime config fail with PostgreSQL version before 9.1
-                    (Reported by Rodrigo Ramirez Norambuena)
- [ASTERISK-27260] - [pjsip] chan_pjsip_indicate: Don't know how to indicate
-                    condition 36 (Reported by Daniel Heckl)
- [ASTERISK-27257] - bridge_native_rtp: half-way direct media when using early
-                    bridging (Reported by Jean Aunis - Prescom)
- [ASTERISK-16898] - SRTP unprotect: authentication failure when RTP sequence
-                    number switches from 65535 -&gt; 0 (Reported by Marcello
-                    Ceschia)
- [ASTERISK-27279] - Crash in pubsub_on_rx_request NULL pointer - Possible
-                    PJSIP Vulnerability (Reported by Ross Beer)
- [ASTERISK-25524] - module reload res_calendar.so does not reload everything
-                    in calendar.conf (Reported by Jesper)
- [ASTERISK-27274] - RTCP needs better packet validation to resist port scans.
-                    (Reported by Richard Mudgett)
- [ASTERISK-27252] - RTP: One way audio with direct media and strictrtp=yes.
-                    (Reported by Richard Mudgett)
- [ASTERISK-24588] - res_calendar does not process CalDAV from Owncloud [fix
-                    included] (Reported by Stefan Gofferje)
- [ASTERISK-25523] - res_calendar: Warning about invalid channel value (for
-                    notification) occurs even when event has no notification
-                    configured. (Reported by Jesper)
- [ASTERISK-21399] - RTP Multicast of L16 (type 10): Asterisk and wireshark
-                    disagree (Reported by Tzafrir Cohen)
- [ASTERISK-27248] - [patch] external_media_address and
-                    external_signaling_address don't always honor localnet
-                    (Reported by Walter Doekes)
- [ASTERISK-27165] - CDR: CDR(start,u) function won't work in cdr_custom config
-                    (Reported by Jacek Konieczny)
- [ASTERISK-24066] - res_smdi: convert to astobj2 (Reported by Corey
-                    Farrell)
- [ASTERISK-27217] - chan_sip: Asterisk crashing when subscription doesn't get
-                    set (Reported by Bryan Walters)
- [ASTERISK-17540] - SDP origin attribute modified when issuing re-INVITE
-                    because of directmedia=yes (Reported by saghul)
- [ASTERISK-27254] - alembic: prune_on_boot fix erroneous (Reported by Florian
-                    Floimair)
- [ASTERISK-27232] - When in queue on g722 with interruptions, music on hold
-                    can get stuck and no longer play (Reported by Jens T.)
- [ASTERISK-27024] - nat/external_media settings ignored in 14.4.1 (Reported by
-                    Christopher van de Sande)
- [ASTERISK-26879] - PJSIP external_media_address ignored if no local_net
-                    options are provided (Reported by Matt Jordan)
- [ASTERISK-27236] - Segfault ast_channel_name (chan=0x0) at
-                    channel_internal_api.c:478 during T.38 Fax Receive
-                    (Reported by Ross Beer)
- [ASTERISK-27225] - Crash when freeing dtls_cfg-&gt;cafile (Reported by Richard
-                    Kenner)
- [ASTERISK-27177] - ooh323c: misleading indentation in
-                    addons/ooh323c/src/ooSocket.c (Reported by Tzafrir Cohen)
- [ASTERISK-27241] - libc segfault upon entry into app_directory (Reported by
-                    David Moore)
- [ASTERISK-27152] - Sending a "tel" uri in a From or To header in an
-                    unauthenticated message causes asterisk to crash (Reported
-                    by Ross Beer)
- [ASTERISK-27103] - core: ast_safe_system command injection possible.
-                    (Reported by Corey Farrell)
- [ASTERISK-27013] - res_rtp_asterisk: Media can be hijacked even with strict
-                    RTP enabled (Reported by Joshua C. Colp)
- [ASTERISK-27231] - res_rtp_asterisk: Allow remote SSRC to change due to
-                    renegotiation (Reported by Joshua C. Colp)
- [ASTERISK-26994] - Confbridge: CBAnn channels intermittently become stuck
-                    when caller hangs up before recording name (Reported by
-                    James Terhune)
- [ASTERISK-27222] - core: Don't queue up multiple video update frames.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20858] - app_minivm fails to clean up mkstemp files (Reported by
-                    Walter Doekes)
- [ASTERISK-16777] - several filename bugs in Record() application (Reported by
-                    klaus3000)
- [ASTERISK-27168] - alembic: PJSIP scripts are missing column dtls_fingerprint
-                    in ps_endpoints table (Reported by Florian Floimair)
- [ASTERISK-27209] - Incorrect SDP in 200 OK when PJSIP_DTMF_MODE is used
-                    (Reported by Torrey Searle)
- [ASTERISK-19103] - When using realtime queues, function QUEUE_MEMBER_LIST()
-                    will return an error if no other app/function has loaded
-                    the queues first. This problem does not exist if
-                    queues.conf is used. (Reported by Jim Van Meggelen)
- [ASTERISK-21241] - When using voicemail as announce only (maxmsg=0), the star
-                    dtmf to enter the voicemail is not honored (Reported by
-                    Eelco Brolman)
- [ASTERISK-27212] - bridge_softmix: Quickly joining/leaving may cause video
-                    stream to remain in SFU (Reported by Richard Mudgett)
- [ASTERISK-27204] - [patch] app_queue: Wrong queue stat calculation (Reported
-                    by Sungtae Kim)
- [ASTERISK-27207] - XMPP OAuth not working due to inverted logic (Reported by
-                    Michael Kuron)
- [ASTERISK-27174] - res_calendar_icalendar: Recurring events not being loaded
-                    from Google calendar using ical (Reported by Mark Thompson)
- [ASTERISK-27202] - If wget is not installed and "or" is not available,
-                    external components (excluding pjsip) are not installed
-                    (Reported by Seán C. McCord)
- [ASTERISK-27200] - manager: hook event is not being raised (Reported by Kevin
-                    Harwell)
- [ASTERISK-27147] - Either asterisk or pjproject isn't re-using tcp
-                    connections (again) (Reported by George Joseph)
- [ASTERISK-27193] - IPv6 receive address in message doesn't include brackets
-                    (Reported by Scott Griepentrog)
- [ASTERISK-27158] - [patch] res_rtp_asterisk: RTCP statistics are not
-                    available when native bridge is used (Reported by Torrey
-                    Searle)
- [ASTERISK-26745] - Asymmetric codecs when asymmetric_rtp_codec=no (Reported
-                    by Jesse Ross)
- [ASTERISK-27189] - Make --with-pjproject-bundled the default for Asterisk 15
-                    (Reported by George Joseph)
- [ASTERISK-27110] - RTP session is not fully destroyed on channel hangup
-                    (Reported by Matt Jordan)
- [ASTERISK-27182] - bridge: Crash when mapping streams (Reported by Joshua C.
-                    Colp)
- [ASTERISK-27180] - channel: requester leaks joint_cap on success. (Reported
-                    by Corey Farrell)
- [ASTERISK-27179] - res_pjsip_session: Handling of 'msid' is incorrect
-                    (Reported by Kevin Harwell)
- [ASTERISK-27119] - res_pjsip: parse/add msid attribute when webrtc is enabled
-                    (Reported by Kevin Harwell)
- [ASTERISK-27171] - Asterisk 15.0.0-Beta1 does not compile (Reported by Ira
-                    Emus)
- [ASTERISK-26659] - res_pjsip: PJSIP presence - missing braces around the
-                    status element in XML (Reported by Abraham Liebsch)
- [ASTERISK-27156] - Asterisk won't compile on Fedora 26 with devmode enabled.
-                    (Reported by Corey Farrell)
- [ASTERISK-27001] - res_pjsip: TLS connection not stable (Reported by Ian
-                    Gilmour)
- [ASTERISK-27130] - Applications ARI: Unsubscribe action for deviceStates does
-                    not remove old subscriptions properly (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-25810] - say.c calls for sounds in the subdir "digits" that don't
-                    exist (in Core). SayUnixTime or other Say... apps will
-                    fail out when they call these sounds. (Reported by Nicolas
-                    Riendeau)
- [ASTERISK-27142] - sounds: Conflict between files in asterisk-sounds-core-1.6
-                    and asterisk-sounds-extra-1.5 (Reported by Corey Farrell)
- [ASTERISK-27143] - bridge_softmix / res_rtp_asterisk: Fix packet loss and
-                    renegotiation issues. (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- -----------------------------------
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-22825] - Dialplan Function for Checking Parking Lot Slot (Reported
-                    by JoshE)
- [ASTERISK-27912] - [PATCH] Add predial handler to app_queue (Reported by
-                    Kristian Høgh)
- [ASTERISK-27929] - [patch] BuildSystem: Enable autotools in Solaris 11.
-                    (Reported by Alexander Traud)
- [ASTERISK-27752] - Ten seconds of silence after mp3 playback (Reported by Sam
-                    Wierema)
- [ASTERISK-27910] - [patch] res_rtp_asterisk: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27906] - [patch] res_crypto: Allow OpenSSL configured with
-                    no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27877] - app_confbridge: Add talking indicator for ConfBridgeList
-                    AMI response (Reported by William McCall)
- [ASTERISK-27873] - documentation: Error on wiki description of Asterisk 13
-                    "MeetmeMute" event (Reported by Alessandro Polidori)
- [ASTERISK-27846] - ast_coredumper: Fix OUTPUT directory (Reported by Ted
-                    G)
- [ASTERISK-27867] - [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured
-                    with no-deprecated. (Reported by Alexander Traud)
- [ASTERISK-27796] - res_hep: Allow create_address to resolve a provided
-                    hostname (Reported by Sebastian Gutierrez)
- [ASTERISK-27820] - [patch] Add DragonFly BSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-25129] - wrong automatic ras address assignment if multihomed
-                    (Reported by Dmitry Melekhov)
- [ASTERISK-27793] - cppcheck identifies redundant "if" (Reported by Ilya
-                    Shipitsin)
- [ASTERISK-27697] - Enable in-dialog NOTIFY on chan_pjsip channels (Reported
-                    by Nathan Bruning)
- [ASTERISK-27770] - [patch] install_prereq: Add Slackware (somehow). (Reported
-                    by Alexander Traud)
- [ASTERISK-27769] - [patch] install_prereq: Add Gentoo Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27738] - [patch] install_prereq: Add Arch Linux. (Reported by
-                    Alexander Traud)
- [ASTERISK-27736] - [patch] install_prereq: Add SUSE. (Reported by Alexander
-                    Traud)
- [ASTERISK-27253] - [patch] libsrtp-2.1.x support (Reported by Alexander
-                    Traud)
- [ASTERISK-27728] - [patch] BuildSystem: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27730] - PJSIP: Update bundled PJPROJECT to version 2.7.2 (Reported
-                    by Richard Mudgett)
- [ASTERISK-27729] - [patch] install_prereq: Add NetBSD. (Reported by Alexander
-                    Traud)
- [ASTERISK-27683] - [patch] BuildSystem: Allow newer autotools on OpenBSD.
-                    (Reported by Alexander Traud)
- [ASTERISK-27348] - [patch] contrib/scripts: add a way to migrate from
-                    chan_sip to chan_pjsip realtime (Reported by Torrey Searle)
- [ASTERISK-27661] - Add new AMI Event for Load, Unload (Reported by Sungtae
-                    Kim)
- [ASTERISK-27651] - app_confbridge: Add Muted to ConfbridgeJoin and channel
-                    snapshot headers to ConfbridgeList AMI events (Reported by
-                    Richard Mudgett)
- [ASTERISK-27647] - app_confbridge/bridge_softmix: When channel muted report
-                    talking stopped if was talking. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27084] - Reduce verbosity while loading PBX extensions. (Reported
-                    by Ludovic Gasc (Eyepea))
- [ASTERISK-24372] - [patch] Add config option to play a prompt to the "winner"
-                    in app_followme (Reported by Graham Mainwaring)
- [ASTERISK-27537] - res_pjsip: Add new AMI Action for PJSIPShowAors (Reported
-                    by Sungtae Kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-24297] - cdr.c: Minor code optimizations. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27470] - Add new object for VoicemailUserEntry (Reported by Sungtae
-                    Kim)
- [ASTERISK-27461] - 3PCC patch for AMI "SIPnotify" (Reported by Yasuhiko
-                    Kamata)
- [ASTERISK-27449] - [PATCH] When failing to acquire target during attended
-                    transfer, display wanted extension (Reported by Niklas
-                    Larsson)
- [ASTERISK-27456] - app_voicemail: Add new object for VoicemailUserEntry
-                    (Reported by Sungtae Kim)
- [ASTERISK-27380] - ast_coredumper: allow pointing out the asterisk binary
-                    explicitly (Reported by Tzafrir Cohen)
- [ASTERISK-23556] - Compilation warning for invert.c (array subscript is above
-                    array bounds) (Reported by Marcello Ceschia)
- [ASTERISK-27359] - pjproject bundled: Don't disable assertions when
-                    --enable-dev-mode is used. (Reported by Corey Farrell)
- [ASTERISK-27355] - Upgrade bundled PJPROJECT to 2.7 (Reported by Richard
-                    Mudgett)
- [ASTERISK-27335] - CDR performance needs improvement. (Reported by Richard
-                    Mudgett)
- [ASTERISK-27278] - [patch] chan_sip: Provide access to read the full SIP
-                    Request-URI from INVITE (Reported by David J. Pryke)
- [ASTERISK-27255] - alembic: Add support for Microsoft SQL server (Reported by
-                    Florian Floimair)
- [ASTERISK-27220] - Enable CHANNEL function to get from and to tag from SIP
-                    Headers (Reported by Andre Nazario)
- [ASTERISK-27169] - Google OAuth 2.0 support for XMPP / Motif (Reported by
-                    Andrey)
- [ASTERISK-27173] - Support for GMIME 3.0 (Reported by Tzafrir
-                    Cohen)
- [ASTERISK-27085] - [patch] chan_pjsip: Port SIPDtmfMode to chan_pjsip
-                    (Reported by Torrey Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="33b1ff03026c100be06d2bfad3fe4000349cdc0d3a58ee91292e23b23fe68d33" name="asterisk" arch="src">
  <version epoch="0" ver="16.5.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.0.1-1" date="1542196800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.6.2 and
- 16.0.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- There is a buffer overflow vulnerability in dns_srv and dns_naptr functions of
- Asterisk that allows an attacker to crash Asterisk via a specially crafted DNS
- SRV or NAPTR response. The attacker’s request causes Asterisk to segfault
- and crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.0.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2018-010.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8c9976f3bb679a7b6492f5011d312be72de508e6664fba94f1a9f63b100ff865" name="asterisk" arch="src">
  <version epoch="0" ver="16.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.0-1" date="1544529600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.1.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on voice packet with marker bit (Reported by
-                    Alexei Gradinari)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                   (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-1" date="1570536000">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9ab6785982c2d676f010eb76a17740c87cbffcc81d7ed5079a0551a3d3caf102" name="asterisk" arch="src">
  <version epoch="0" ver="16.6.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.1.1-1" date="1545825600">- The Asterisk Development Team would like to announce the release of Asterisk 16.1.1.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.1.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.1.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-1" date="1570536000">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-2" date="1571054400">- Fixed systemd service script</changelog>
</package>
<package pkgid="46364ad6c1bb50aba9805dce4b1fd87c531dafeafcdffb2d140ca9e0ae8023bf" name="asterisk" arch="src">
  <version epoch="0" ver="16.6.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.0-1" date="1550232000">- The Asterisk Development Team would like to announce the release of Asterisk 16.2.0.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
-
- For a full list of changes in this release, please see the ChangeLog:
- http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-1" date="1570536000">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-2" date="1571054400">- Fixed systemd service script</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.1-1" date="1571227200">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b477e7ed6e8ba49b7aef42d74d672dee0017a1a0d6161f52e30c66899e611652" name="asterisk" arch="src">
  <version epoch="0" ver="16.6.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.2.1-1" date="1551355200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 15 and 16. The available releases are released as versions 15.7.2 and
- 16.2.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-001: Remote crash vulnerability with SDP protocol violation when
-               Asterisk makes an outgoing call, a very specific SDP protocol
-               violation by the remote party can cause Asterisk to crash.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.2.1
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-001.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.3.0-1" date="1554379200">- The Asterisk Development Team would like to announce the release of Asterisk 16.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-28314] - ARI: API changed but "apiVersion" in rest-api\resources.json
-                    did not (Reported by Stefan Repke)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.0-1" date="1559217600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.4.1-1" date="1562846400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16, and Certified Asterisk 13.21. The available releases are
- released as versions 13.27.1, 15.7.3, 16.4.1 and 13.21-cert4.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-002: Remote crash vulnerability with MESSAGE messages a specially
-               crafted SIP in-dialog MESSAGE message can cause Asterisk to
-               crash.
- AST-2019-003: Remote Crash Vulnerability in chan_sip channel driver when T.38
-               faxing is done in Asterisk a T.38 reinvite may be sent to an
-               endpoint to switch it to T.38. If the endpoint responds with an
-               improperly formatted SDP answer including both a T.38 UDPTL
-               stream and an audio or video stream containing only codecs not
-               allowed on the SIP peer or user a crash will occur. The code
-               incorrectly assumes that there will be at least one common codec
-               when T.38 is also in the SDP answer.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.27.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.4.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert4
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.0-1" date="1564056000">- The Asterisk Development Team would like to announce the release of Asterisk 16.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.5.1-1" date="1567684800">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 15 and 16. The available releases are released as versions
- 13.28.1, 15.7.4 and 16.5.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2019-004: Crash when negotiating for T.38 with a declined stream When
-                 Asterisk sends a re-invite initiating T.38 faxing, and the
-                 endpoint responds with a declined media stream a crash will
-                 then occur in Asterisk.
-
- * AST-2019-005: Remote Crash Vulnerability in audio transcoding When audio
-                 frames are given to the audio transcoding support in Asterisk
-                 the number of samples are examined and as part of this a
-                 message is output to indicate that no samples are present. A
-                 change was done to suppress this message for a particular
-                 scenario in which the message was not relevant. This change
-                 assumed that information about the origin of a frame will
-                 always exist when in reality it may not.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.28.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-15.7.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.5.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-1" date="1570536000">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.0-2" date="1571054400">- Fixed systemd service script</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.1-1" date="1571227200">- The Asterisk Development Team would like to announce the release of Asterisk 16.6.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.6.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.6.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.1-2" date="1571832000">- Added res_json module</changelog>
</package>
<package pkgid="6aabf32eda351f4d31dc7676f066a2d99ef8a5b907a7d27217797e8b74935b3f" name="asterisk" arch="src">
  <version epoch="0" ver="16.6.2" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.1-1" date="1551441600">- Update to upstream 16.2.1 release for security / CVE-2019-7251 / AST-2019-001</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cc589c91f9e11598307869a65c7e7814c72c69873f8b69ea54bad2cd05619177" name="asterisk" arch="src">
  <version epoch="0" ver="16.7.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="99094ac2f6d2e7ff9a31dc88cf12024fb15432869d2466c640f07009d08cccbc" name="asterisk" arch="src">
  <version epoch="0" ver="16.8.0" rel="1.el7_9"/>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="8ad748fe86194c8b61acb3a431c4941cfac67aad4bdc883fffa89df102098227" name="asterisk" arch="src">
  <version epoch="0" ver="16.9.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.6.2-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.7.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.8.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 16.9.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 16.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 16.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl) 
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes) 
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="0b4474d0c8abc08c08fd9614a79bbf91608fd34e70bfaf12cc17786bcd3158a0" name="asterisk" arch="src">
  <version epoch="0" ver="17.0.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.2.1-1" date="1551441600">- Update to upstream 16.2.1 release for security / CVE-2019-7251 / AST-2019-001</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d35fd353dc476baa85c8b9beb959513dff292c0adc72d8a0ec30799f1965a77a" name="asterisk" arch="src">
  <version epoch="0" ver="17.0.1" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.0-1" date="1559304000">- Update to upstream 16.4.0 release for bug fixes</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="7e8d92a429ba78b37d9f3e04e1ff4b123e5755d36dc44b5481640bcc14a6e8ee" name="asterisk" arch="src">
  <version epoch="0" ver="17.1.0" rel="1.el7_9"/>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 16.4.0-2" date="1559390400">- Perl 5.30 rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="21305728506bbe5ec699a6f1fffa551d9ea18db260e64927b10454c39c53dab8" name="asterisk" arch="src">
  <version epoch="0" ver="17.2.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.4.1-1" date="1563192000">- Update to upstream 16.4.1 release for security updates AST-2019-002 and
  AST-2019-003 related to remote crash vulnerabilities</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="868ba37ae55156ddbfa1f2d5cfc167822951b904bc48986ccbbb5f19d4de5ee8" name="asterisk" arch="src">
  <version epoch="0" ver="17.3.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 16.4.1-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul 
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix) 
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="13583bd39d2b7f1409f6ea27b89342f3d4b64e2c5e884fa41ce22a359a8a2eed" name="asterisk" arch="src">
  <version epoch="0" ver="17.4.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.0-1" date="1564056000">- Update to upstream 16.5.0 release for security and bug fixes</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d4b741363412938d508203c4fc3c4c6d45ce570b449533583c7f83e709d2cf39" name="asterisk" arch="src">
  <version epoch="0" ver="17.5.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.5.1-1" date="1567771200">- Update for upstream security release 16.5.1, with AST-2019-004 and
  AST-2019-005</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="bdbd1cff56eb300bdaa3afb0417c9beeb2badb350fafe57d667d81678fbd3253" name="asterisk" arch="src">
  <version epoch="0" ver="17.5.1" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.0-1" date="1570622400">- Update to upstream 16.6.0 for security and bug fixes
- Update to using bundled pjproject release 2.9</changelog>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d85ed1a8a0e94d36fa0924e1fd9e74c874d19f98dfe843566aa277817202fa95" name="asterisk" arch="src">
  <version epoch="0" ver="17.6.0" rel="1.el7_9"/>
  <changelog author="Jared K. Smith &lt;jsmith@fedoraproject.org&gt; - 16.6.1-1" date="1571400000">- Update to upstream 16.6.1 for bug fixes
- Work on building in EPEL-7 and EPEL-8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ca501d124c9bbad912d0a03b1d3662b1643c5ed9687e8cbe0b73b81eea0408bd" name="asterisk" arch="src">
  <version epoch="0" ver="17.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.0-1" date="1572264000">- The Asterisk Development Team would like to announce the release of Asterisk 17.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
- [ASTERISK-28447] - res_pjsip_messaging: In-dialog MESSAGE with no body causes
-                    crash (Reported by Gil Richard)
- [ASTERISK-28465] - Broken SDP can cause a segfault in a T.38 reINVITE (Reported
-                    by Francesco Castellano)
- [ASTERISK-28260] - Asterisk segfault when rtp negotiation is wrong or fails
-                    (Reported by Sotiris Ganouris)
- [ASTERISK-28127] - Buffer overflow for DNS SRV/NAPTR records (Reported by Jan
-                    Hoffmann)
- [ASTERISK-28013] - res_http_websocket: Crash when reading HTTP Upgrade requests
-                    (Reported by Sean Bright)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28403] - Add native Prometheus support to Asterisk (Reported by Matt
-                    Jordan)
- [ASTERISK-28375] - res_pjsip: New configuration setting to allow disabling
-                    norefersub (Reported by Dan Cropp)
- [ASTERISK-28320] - Added ARI resource /ari/channels/{channelid}/rtp_statistics
-                    (Reported by sungtae kim)
- [ASTERISK-28267] - res_stasis: Add ability to switch applications (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-28087] - add flag to allow CALLERID(num) to be placed in Contact
-                    header in chan_pjsip (Reported by Torrey Searle)
- [ASTERISK-27971] - res_pjsip: Implement additional SIP RFCs for Google Voice
-                    trunk compatability (Reported by Nick French)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28477] - Crash when not specifying "dbfile" in
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28478] - Crash performing "core reload" with modified
-                    res_config_sqlite3.conf (Reported by Dennis)
- [ASTERISK-28282] - AST_SCHED_REPLACE_UNREF causes wait-on-self deadlocks (in
-                    chan_sip) (Reported by Walter Doekes)
- [ASTERISK-27121] - res_pjsip_mwi: Memory leak on reload (Reported by Sergej
-                    Kasumovic)
- [ASTERISK-28457] - [patch] Fix crash in chan_dahdi on 32-bit systems caused by
-                    ASTERISK-28317 (Reported by abelbeck)
- [ASTERISK-28458] - res_pjsip_sdp_rtp: Remove unused variable (Reported by
-                    Michael Maier)
- [ASTERISK-26006] - Show offending IP for TLS setup failures in logs (Reported
-                    by Oleksandr Natalenko)
- [ASTERISK-28444] - chan_pjsip: Peer IP for SSL handshake errors not logged
-                    (Reported by Bernhard Schmidt)
- [ASTERISK-26968] - chan_pjsip: Transfer() does not result in TRANSFERSTATUS
-                    reflecting SIP response to transfer (Reported by Dan Cropp)
- [ASTERISK-28419] - app_amd: Does not work with silence suppression (Reported by
-                    Nasir Iqbal)
- [ASTERISK-28018] - IP Fragmentation happening instead of DTLS fragmentation on
-                    handshake server hello certificate (Reported by vijay kumar)
- [ASTERISK-25371] - Crash in hangup at chan_pjsip.c:1749 when Asterisk attempts
-                    to generate hangup event (Reported by Abhay Gupta)
- [ASTERISK-28435] - cdr_pgsql: Unix socket doesn't work (Reported by Dmitry
-                    Svyatogorov)
- [ASTERISK-27981] - res_fax: Fax session leak with fax gatewaying (Reported by
-                    pasandev)
- [ASTERISK-28427] - new mwi.h include missing from some dahdi source files,
-                    causes build failure (Reported by Guido Falsi)
- [ASTERISK-28421] - Wrong type used for timestamp in res_rtp_asterisk (Reported
-                    by Morten Tryfoss)
- [ASTERISK-27994] - PJSIP: Early media ringback not indicated after Progress()
-                    (Reported by Gregory Massel)
- [ASTERISK-28412] - GCC 9 catches more string formatting issues (Reported by
-                    George Joseph)
- [ASTERISK-28379] - pjsip: show channelstats incorrect information output
-                    (Reported by Vyrva Igor)
- [ASTERISK-28399] - channel.c: Exceptionally long queue length queuing (Reported
-                    by Abhay Gupta)
- [ASTERISK-28392] - The no-partial-inlining flag isn't passed to the bundled
-                    pjproject or jansson builds (Reported by George Joseph)
- [ASTERISK-28402] - res_pjsip_registrar: SEGV in registrar_find_contact
-                    (Reported by Ross Beer)
- [ASTERISK-27756] - bridge: Failure to impart a channel results in bad data
-                    causing crash (Reported by Abhay Gupta)
- [ASTERISK-26718] - ARI: Bridge destroying doesn't work as expected (Reported by
-                    Marin Odrljin)
- [ASTERISK-28143] - app_amd: Infinite loop on silent calls (Reported by Abhay
-                    Gupta)
- [ASTERISK-28353] - stasis: Crash at shutdown when statistics enabled (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28374] - latest asterisk unconditionally launch gcc --version, even
-                    if the compiler is different (Reported by Guido Falsi)
- [ASTERISK-28391] - res_indications: Crash requesting autocomplete on
-                    indications cli command (Reported by Lucas Mendes)
- [ASTERISK-27935] - app_voicemail: emailbody per user can't contain commas
-                    (Reported by Sébastien Duthil)
- [ASTERISK-17695] - 1.8.3.2 extenpatternmatchnew=yes cannot find extensions with
-                    '-' in them (Reported by test011)
- [ASTERISK-17799] - AEL reload causes loss of control in a macro (Reported by
-                    Kirill Katsnelson)
- [ASTERISK-18593] - AEL for loops use Macro app and pipe delimiter (Reported by
-                    Luke-Jr)
- [ASTERISK-14939] - AEL parsers does not find existing label (Reported by
-                    klaus3000)
- [ASTERISK-20182] - Parsing a label beginning with a numeric character in all
-                    Goto/GotoIf/GotoIfTime application causes unexpected
-                    behavior (Reported by Janu)
- [ASTERISK-28348] - Failed to initialize OOH323 endpoint-OOH323 Disabled
-                    (Reported by Dmitry Shubin)
- [ASTERISK-28371] - chan_pjsip: DTMF Mode auto_info fallback lead to both inband
-                    and info (Reported by Salah Ahmed)
- [ASTERISK-28319] - musl: Crash on startup when loading modules (Reported by
-                    Sebastian Kemper)
- [ASTERISK-28362] - strtok_r() makes gcc compile warning (Reported by sungtae
-                    kim)
- [ASTERISK-28255] - res_rtp_asterisk: REMB RTCP packet sending may be incorrect
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27541] - app_queue: Queue paused reason was (big number) secs ago
-                    when reason is set (Reported by César Benjamín García
-                    Martínez)
- [ASTERISK-20986] - QUEUE_MEMBER 's description is inaccurate (Reported by
-                    Olivier Krief)
- [ASTERISK-28350] - manager: Stasis backed up due to locking (Reported by Joshua
-                    C. Colp)
- [ASTERISK-25792] - chan_sip: qualifygap bounds checking (Reported by Paul
-                    Sandys)
- [ASTERISK-28341] - res_config_odbc eliminates empty custom (“@” prefix)
-                    variables (Reported by Alexei Gradinari)
- [ASTERISK-28333] - StasisEnd event makes wrong timestamp value (Reported by
-                    sungtae kim)
- [ASTERISK-28306] - res_pjsip_mwi: MWI NOTIFY occasionally takes minutes to be
-                    sent (Reported by Jared Hull)
- [ASTERISK-28332] - Variable ALTCONF ignored when service is used in Debian
-                    (Reported by Cirillo Ferreira)
- [ASTERISK-27964] - app_queue: ring_entry accesses nativeformats without channel
-                    lock or reference (Reported by Francisco Seratti)
- [ASTERISK-28335] - stasis: Make topic and maybe subscription names unique and
-                    more useful (Reported by Joshua C. Colp)
- [ASTERISK-28321] - res_rtp_asterisk: Fixing possible divide by zero for rtcp
-                    stat calculation (Reported by sungtae kim)
- [ASTERISK-28322] - chan_pjsip: Add option to allow ignoring of 183 without SDP
-                    (Reported by Torrey Searle)
- [ASTERISK-28328] - MeetMe global non-admin mute is muting admins that
-                    subsequently join (Reported by Philip Mott)
- [ASTERISK-28168] - app_queue: Adding a blank entry into sql queue_members
-                    crashes asterisk. (Reported by Michael)
- [ASTERISK-28323] - pjsip: sip.conf to pjsip.conf conversion script fails
-                    (Reported by Guido Weckwerth)
- [ASTERISK-28272] - The basic-pbx config samples don't produce a running
-                    asterisk (Reported by George Joseph)
- [ASTERISK-28312] - res_pjsip_diversion: Corrupted SIP Diversion field after
-                    handling a 302 redirect (Reported by Alex Odrov)
- [ASTERISK-24173] - File menuselect/menuselect_gtk.c has no license header
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28166] - app_voicemail: Asterisk unresponsive after changing
-                    voicemail password with ODBC (Reported by Michael)
- [ASTERISK-28309] - res_pjsip: Wrong Contact and Via fields with multiple UDP
-                    interfaces (Reported by Nikolay shakin)
- [ASTERISK-27992] - PJSIP: Adding `sends_registrations = yes` to
-                    pjsip_wizard.conf causes crash (Reported by Jonathan Harris)
- [ASTERISK-28213] - res_pjsip: Threads pile up needlessly when AOR is blocked
-                    (Reported by Ross Beer)
- [ASTERISK-28301] - Allow voicemail boxes to be subscribed to with a presence
-                    event package (Reported by George Joseph)
- [ASTERISK-28303] - res_rtp_asterisk: Interaction between smoother and DTMF can
-                    cause out of order timestamps (Reported by Torrey Searle)
- [ASTERISK-28302] - ARI: "Error destroying mutex" when listing all ARI
-                    applications (Reported by Stefan Repke)
- [ASTERISK-28300] - AST_PBX_MAX_STACK is too low for some applications (Reported
-                    by George Joseph)
- [ASTERISK-28106] - Astricon Feedback: Unable to filter ARI events when GETting
-                    causes overload of events (Reported by George Joseph)
- [ASTERISK-28284] - switching between native_bridge and simple_bridge can cause
-                    one way audio (Reported by Torrey Searle)
- [ASTERISK-28251] - CI: Fix CI so it reverifies commit message changes (Reported
-                    by George Joseph)
- [ASTERISK-28277] - database: Add some basic logging (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28181] - ari: Originating overwrites channel start time (Reported by
-                    sungtae kim)
- [ASTERISK-28173] - Deadlock in chan_sip handling subscribe request during
-                    res_parking reload (Reported by Giuseppe Sucameli)
- [ASTERISK-28104] - AstriCon Feedback: Automatically create a 1 line dialplan
-                    context for stasis apps (Reported by George Joseph)
- [ASTERISK-28271] - Opensuse Leap 15 --with-jannson-bundled will not compile
-                    (Reported by David Wilcox)
- [ASTERISK-28238] - PJSIP realtime. getcontext not working with DUNDI (Reported
-                    by Ray)
- [ASTERISK-28263] - codec_opus: errors setting max_playback_rate and bitrate to
-                    "sdp" (Reported by Gianluca Merlo)
- [ASTERISK-28257] - res_http_websocket: PING / PONG opcodes break data reception
-                    (Reported by Jeremy Lainé)
- [ASTERISK-28250] - build: Cross-compilation fails for target
-                    arm-linux-gnueabihf (Reported by Jean Aunis - Prescom)
- [ASTERISK-28252] - HangupHandler manager events are never thrown (Reported by
-                    Gerald Schnabel)
- [ASTERISK-28231] - res_http_websocket: Not responding to Connection Close Frame
-                    (opcode 8) (Reported by Jeremy Lainé)
- [ASTERISK-28249] - res_monitor: Segfault with Monitor(wav,file,i) (Reported by
-                    Valentin Vidić)
- [ASTERISK-28244] - stasis: Filter messages at publishing to AMI/ARI (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28197] - stasis: ast_endpoint struct holds the channel_ids of
-                    channels past destruction in certain cases (Reported by
-                    Mohit Dhiman)
- [ASTERISK-28230] - res_rtp_asterisk: abs-send-time extension added with
-                    Asterisk 15.5.0 breaks GXV3140 video telephony (Reported by
-                    David Kuehling)
- [ASTERISK-28232] - core: RAII using clang use-after-scope issue (Reported by
-                    Diederik de Groot)
- [ASTERISK-28162] - [patch] need to reset DTMF last sequence number and
-                    timestamp on RTP renegotiation (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28225] - app_voicemail: Channel variable VM_MESSAGEFILE not updated
-                    correctly if message marked "urgent" (Reported by boatright)
- [ASTERISK-28218] - app_queue: Asterisk crashes when using Queue with a pre-dial
-                    handler (option b) (Reported by Mark)
- [ASTERISK-28212] - stasis: Statistics broke ABI under developer mode (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28222] - Regression: MWI polling no longer works (Reported by
-                    abelbeck)
- [ASTERISK-28221] - Bug in ast_coredumper (Reported by Andrew Nagy)
- [ASTERISK-28215] - app_voicemail: Leaving voicemail sometimes doesn't trigger
-                    NOTIFYs (Reported by George Joseph)
- [ASTERISK-27959] - [patch] Asterisk 15.4.1 h264 fmtp negotiation problem
-                    (Reported by David Kuehling)
- [ASTERISK-28201] - [patch] confbridge: no announce to the marked users when
-                    they join an empty conference (Reported by Alexei Gradinari)
- [ASTERISK-28117] - stasis: Add statistics for usage when in developer mode
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28186] - stasis: Filter messages at publishing based on to_* presence
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28194] - chan_sip: Leak using contact ACL (Reported by Giuseppe
-                    Sucameli)
- [ASTERISK-28157] - Asterisk crashes when the res_pjsip_* modules unload
-                    (Reported by sungtae kim)
- [ASTERISK-28125] - app_queue: Revert broken queue channel reference patch
-                    (Reported by lvl)
- [ASTERISK-27095] - chan_pjsip: When connected_line_method is set to invite,
-                    we're not trying UPDATE (Reported by George Joseph)
- [ASTERISK-28182] - chan_pjsip: When connected_line_method is set to invite,
-                    asterisk is not trying UPDATE (Reported by nappsoft)
- [ASTERISK-28151] - app_voicemail: MWI fails with mailboxes=##@device instead of
-                    mailboxes=##@default (Reported by Ronald Raikes)
- [ASTERISK-28119] - stasis: Segment channel snapshot to reduce creation cost
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28102] - stasis: Use implementation specific cache for channel
-                    snapshots (Reported by Joshua C. Colp)
- [ASTERISK-28159] - SIGABRT caused by stack corruption in hashkeys_read when no
-                    matching keys present (Reported by Michael Walton)
- [ASTERISK-28140] - repeated segmentation faults (Reported by Eyal Hasson)
- [ASTERISK-28103] - stasis: Filter messages at publishing to reduce work done
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28169] - ARI /channels/create handler causes core dump (Reported by
-                    sungtae kim)
- [ASTERISK-28129] - Incorrect Behavior for rewrite_contact when Re-Invite omits
-                    routset (Reported by Torrey Searle)
- [ASTERISK-28158] - Some conditions prevent running of el_end, break the
-                    terminal. (Reported by Corey Farrell)
- [ASTERISK-28110] - rtp: Incorrect Packetization (Reported by Robert Cripps)
- [ASTERISK-28146] - pbx_config: Only the first [globals] section is processed.
-                    (Reported by Corey Farrell)
- [ASTERISK-28150] - Formatting error in documentation (Reported by Scott
-                    Griepentrog)
- [ASTERISK-28081] - chan_sip: Asterisk 12+ chan_sip doesn't report
-                    AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit
-                    van Drongelen)
- [ASTERISK-28137] - res_pjsip_notify: improve realtime performance on CLI
-                    completion on the endpoint (Reported by Alexei Gradinari)
- [ASTERISK-27980] - Caller ID cannot be changed on Attended Transfer before
-                    dialing out (Reported by Alexei Gradinari)
- [ASTERISK-28107] - app_confbridge: Participant info labels aren't being added
-                    to the SDPs (Reported by George Joseph)
- [ASTERISK-28089] - function ast_sendtext() create RTP realtime packets with a
-                    trailing null byte in the payload (Reported by Emmanuel BUU)
- [ASTERISK-28076] - bridging: Asterisk crashes when receiving an empty realtime
-                    text frame (Reported by Emmanuel BUU)
- [ASTERISK-28084] - app_queue: QueueMemberStatus Event flooding AMI (Reported by
-                    Andrej)
- [ASTERISK-28077] - res_pjsip: improve realtime performance on CLI 'pjsip show
-                    contacts' (Reported by Alexei Gradinari)
- [ASTERISK-27920] - app_queue: Queue member considered inuse after immediately
-                    hanging up during dialing. (Reported by Cao Minh Hiep)
- [ASTERISK-26094] - stasis: Playing MOH to bridge with ARI does not work
-                    (Reported by Cameron)
- [ASTERISK-28065] - res_odbc: missing SQL error diagnostic (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28057] - chan_sip: SipNotify via AMI behaves differently to CLI
-                    (Reported by Peter Katzmann)
- [ASTERISK-28045] - configure script does not enforce libunbound2 version
-                    (Reported by Samuel Galarneau)
- [ASTERISK-28070] - testsuite: Sniffer assumes pjmedia will use ports below
-                    10000 (Reported by Joshua C. Colp)
- [ASTERISK-27854] - rtp: Crash in off-nominal case where RTP instance can't be
-                    set up (Reported by Lei Fu)
- [ASTERISK-28034] - chan_sip unstable with TLS after asterisk start or reloads
-                    (Reported by David Hajek)
- [ASTERISK-28059] - PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28047] - chan_pjsip: Declined video stream is added when no video
-                    codecs configured and session refresh with removed video
-                    stream occurs (Reported by Will)
- [ASTERISK-28033] - AMI event "NewExten" is set to the wrong class (Reported by
-                    lvl)
- [ASTERISK-28049] - res_pjproject build failure (Reported by Jaco Kroon)
- [ASTERISK-28029] - [patch] res_musiconhold : music on hold will not start if
-                    previous hold just reached end of file (Reported by Frederic
-                    LE FOLL)
- [ASTERISK-28005] - channel.c: ARI ring only once (Reported by Hajek Michal)
- [ASTERISK-28032] - Realtime queuemembers are not updated during retry phase
-                    (Reported by lvl)
- [ASTERISK-27988] - alembic: PJSIP "mwi_subscribe_replaces_unsolicited" field is
-                    integer not boolean (Reported by Joshua C. Colp)
- [ASTERISK-28020] - res_pjsip_transport_websocket: Properly set 'received' for
-                    IPv6 (Reported by Sean Bright)
- [ASTERISK-28002] - When T.140 realtime text is negociated, a lot of debug
-                    traces are generated (Reported by Emmanuel BUU)
- [ASTERISK-27881] - PBX calls via chan_sip TCP trunk now get authentification
-                    error (Reported by Ian Gilmour)
- [ASTERISK-28022] - res_pjsip realtime: uri column in ps_contacts table can be
-                    too short (Reported by Florian Floimair)
- [ASTERISK-27944] - res_pjsip_t38: Crash receiving 1xx responses other than 100
-                    before 200 for T.38 reINVITE (Reported by Joshua Elson)
- [ASTERISK-28007] - rtcp-mux is put in SDP answer regardless of offer (Reported
-                    by Torrey Searle)
- [ASTERISK-27398] - No joint capabilities with video and audio-only streams
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-27973] - app_queue: QUEUESTATUS = CONTINUE instead LEAVEEMPTY
-                    (Reported by Valentin Safonov)
- [ASTERISK-27997] - pjproject_bundled: Fix for Solaris builds. Do not undef
-                    s_addr. (Reported by Alexander Traud)
- [ASTERISK-27999] - Wrong SRTP use status report (Reported by Salah Ahmed)
- [ASTERISK-28001] - res_pjsip_registrar: Improve performance of inbound handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-27966] - pjsip: Race condition in 183 re transmission can result in a
-                    deadlock (Reported by Torrey Searle)
- [ASTERISK-15331] - make menuselect fails due to undefined symbols (initscr32,
-                    w32addch) in menuselect_curses.o (Reported by Majdi Bsoul)
- [ASTERISK-14935] - [regression] menuselect compilation failure on Solaris 10
-                    (Reported by Samuel Owens)
- [ASTERISK-12382] - menuselect compilation failure on Solaris 10 / gcc 3.4.3
-                    (Reported by rleasure)
- [ASTERISK-9107] - menuselect compilation failure on Solaris 10/gcc-4.1.1
-                    (Reported by Bob Atkins)
- [ASTERISK-27991] - BuildSystem: Enable Jansson in Solaris 11. (Reported by
-                    Alexander Traud)
- [ASTERISK-27548] - res_pjsip_endpoint_identifier_ip only matches against
-                    "generic string" headers (Reported by George Joseph)
- [ASTERISK-27990] - res_rtp_asterisk: Requires OpenSSL in Developer Mode.
-                    (Reported by Alexander Traud)
- [ASTERISK-27591] - Frack errors in stasis.c and memory leakage (Reported by
-                    Siruja Maharjan)
- [ASTERISK-27978] - res_pjsip: Change default transport keepalive to preserve
-                    behavior (Reported by Joshua C. Colp)
- [ASTERISK-27968] - systemd: asterisk.service (Reported by seanchann.zhou)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28443] - app_voicemail: remove dependency on stasis cache (Reported
-                    by Kevin Harwell)
- [ASTERISK-28442] - stasis_state: Create a stasis module to cache last known
-                    state (Reported by Kevin Harwell)
- [ASTERISK-28385] - res_ari_channels: Added detail hangup code settings
-                    (Reported by sungtae kim)
- [ASTERISK-28234] - pbx_dundi: Add IPv4/IPv6 dual bind support for DUNDi
-                    (Reported by Kirsty Tyerman)
- [ASTERISK-28401] - app_confbridge: Add *_all remb behavior variants (Reported
-                    by Joshua C. Colp)
- [ASTERISK-28400] - res_rtp_asterisk / res_pjsip_sdp_rtp: Add support for
-                    transport-cc (Reported by Joshua C. Colp)
- [ASTERISK-28363] - Millisecond-resolution call stats including PDD in channel
-                    variables (Reported by Antoni Goldstein)
- [ASTERISK-28378] - Added detail subscriber/subscription info for stasis show
-                    app cli (Reported by sungtae kim)
- [ASTERISK-20207] - Asterisk should clear out any .lock files in the voice mail
-                    directory on startup. (Reported by Steven Wheeler)
- [ASTERISK-28111] - build: CHANGES/UPGRADE are irritating to work with.
-                    (Reported by Corey Farrell)
- [ASTERISK-28264] - Added topic_all container (Reported by sungtae kim)
- [ASTERISK-28343] - Added app_name, app_data to channel type (Reported by
-                    sungtae kim)
- [ASTERISK-28326] - ari: Added timestamp for some ari events. (Reported by
-                    sungtae kim)
- [ASTERISK-28317] - Add logical group at DAHDIChannel event and create
-                    "dahdi_group" at CHANNEL function (Reported by Cirillo
-                    Ferreira)
- [ASTERISK-28279] - Added creation timestamp for bridge (Reported by sungtae
-                    kim)
- [ASTERISK-27483] - Allow wrapuptime to be set for each queue member (Reported
-                    by Rodrigo Ramirez Norambuena)
- [ASTERISK-28055] - app_queue: Per-member wrapup time missing from
-                    AddQueueMember application (Reported by Niksa Baldun)
- [ASTERISK-28292] - Changed to show all channel stats including wrong media
-                    (Reported by sungtae kim)
- [ASTERISK-28253] - res_pjsip_session: Adding rtcp stats result into the session
-                    (Reported by sungtae kim)
- [ASTERISK-28246] - Support skipping on the g726 format (Reported by Eyal
-                    Hasson)
- [ASTERISK-28196] - bridge_softmix: Does not support WebRTC source with multi
-                    video tracks. (Reported by Xiemin Chen)
- [ASTERISK-28198] - res_ari: Add new hangup causes for ARI Channel DELETE
-                    command (Reported by Sebastian Damm)
- [ASTERISK-28144] - [patch] New function PJSIP_PARSE_URI to parse an URI and
-                    return a specified part of the URI (Reported by Alexei
-                    Gradinari)
- [ASTERISK-28136] - Allow the sip_to_pjsip script to be used in a pipe (Reported
-                    by Pascal Cadotte Michaud)
- [ASTERISK-28046] - Remove stale nonoptreq references (Reported by Walter
-                    Doekes)
- [ASTERISK-27164] - [patch] Add IPv6 Support for DUNDi (Reported by Adam
-                    Secombe)
- [ASTERISK-28006] - PJSIP: Missing "party=calling"/"party=called" in
-                    Remote-Party-ID (Reported by Eric Dantie)
- [ASTERISK-27995] - pjproject_bundled: Find shared libraries in root
-                    --with-ssl=PATH. (Reported by Alexander Traud)
- [ASTERISK-27993] - pjsip_wizard example gives wrong info about unsupported SRV
-                    records (Reported by Jonathan Harris)
- [ASTERISK-27970] - res_rtp_asterisk: T.140 packets containing backspace or end
-                    of line are merged with regular text and it causes some UA
-                    to break (Reported by Emmanuel BUU)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f1d4d6f23c2d8b2988e3187b5b7784da1f13761c502c1e14071168082aec285e" name="asterisk" arch="src">
  <version epoch="0" ver="17.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.0.1-1" date="1574337600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16 and 17, and Certified Asterisk 13.21. The available releases are
- released as versions 13.29.2, 16.6.2, 17.0.1 and 13.21-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2019-006: SIP request can change address of a SIP peer. A SIP request can
-               be sent to Asterisk that can change a SIP peer’s IP address. A
-               REGISTER does not need to occur, and calls can be hijacked as a
-               result. The only thing that needs to be known is the peer’s name;
-               authentication details such as passwords do not need to be known.
-               This vulnerability is only exploitable when the “nat” option is
-               set to the default, or “auto_force_rport”.
- AST-2019-007: AMI user could execute system commands. A remote authenticated
-               Asterisk Manager Interface (AMI) user without “system”
-               authorization could use a specially crafted “Originate” AMI
-               request to execute arbitrary system commands.
- AST-2019-008: Re-invite with T.38 and malformed SDP causes crash. If Asterisk
-               receives a re-invite initiating T.38 faxing and has a port of 0
-               and no c line in the SDP, a crash will occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.29.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.6.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-13.21-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2019-006.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2019-008.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e19f99ec93792b91b0eb443bcdb6793191d08f45c11bc37861fb4d0096fec6d6" name="asterisk" arch="src">
  <version epoch="0" ver="17.8.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="211744e2385a8a19defd57335f6526af2c7db68754c76084efaebc63db1808f7" name="asterisk" arch="src">
  <version epoch="0" ver="17.9.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cc386223903c0b31f628f432739cb76234604caade1eeaadf6f8b47095c16274" name="asterisk" arch="src">
  <version epoch="0" ver="17.9.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="db0cddbce7d7c7e585303848c696411374e61bc381c87f1b299379112823e75f" name="asterisk" arch="src">
  <version epoch="0" ver="17.9.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="fefbe1104b565521a8355e2c325ce6908a879ec37c72277183eee0d543a0784a" name="asterisk" arch="src">
  <version epoch="0" ver="17.9.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="91ba649e925cf4d9e5038cf28e69cc4cdf9862a3ab06865d9bb6b870dc2c3277" name="asterisk" arch="src">
  <version epoch="0" ver="17.9.4" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="176cf600fd34dde357d29c14e2b31ed3b93b4176f8e6b4fde7b81514561ca7de" name="asterisk" arch="src">
  <version epoch="0" ver="17.9.4" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-2" date="1627992000">- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="ef595fd0d1c9919d752debc12aa4c14dcf9db6d1933fad0af198e177871a48c2" name="asterisk" arch="src">
  <version epoch="0" ver="17.9.4" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 17.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.2-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.3-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.9.4-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
</package>
<package pkgid="3e07f3e33c3055b7c1c19135df3511f658bdfd0521a8f01ed816cb8d0747c4e8" name="asterisk" arch="src">
  <version epoch="0" ver="18.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.1.0-1" date="1577102400">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.1.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- -----------------------------------
-
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup.
-                    (Reported by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported by
-                    Florian Floimair)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime.
-                    (Reported by Frederic LE FOLL)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout (Reported
-                    by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when target
-                    line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by Aheliotech)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin Harwell)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril Ramière)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported by
-                    Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey Searle)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="dbd4acdfeef585112b7960a4ba94a1cb816d520827cee758027a4ac99f0b3eb7" name="asterisk" arch="src">
  <version epoch="0" ver="18.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.2.0-1" date="1580817600">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation' output
-                    (Reported by Sean Bright)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch] Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by
-                    Stas Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event (Reported
-                    by Niksa Baldun)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not preserve
-                    XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported by
-                    Boris P. Korzun)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media stream
-                    if codecs create additional streams and offer does not have
-                    them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="18e4456be60d67caa10e81236c3fe8b1ed118d30936bf4ad8e7e1ee9e3ed30e0" name="asterisk" arch="src">
  <version epoch="0" ver="18.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.3.0-1" date="1584014400">- The Asterisk Development Team would like to announce the release of Asterisk 17.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28719] - Cannot remove defaultrule from queue using realtime queues
-                    (Reported by EDV O-TON)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation (Reported
-                    by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5a24e4bdc9d2a5812a6a3e7dd471b7b753d4f812a39f8b45d54de390e6c60442" name="asterisk" arch="src">
  <version epoch="0" ver="18.1.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.4.0-1" date="1588248000">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.4.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging for
-                    a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP and
-                    TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio stream
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not consistent
-                    with examples. Missing examples. (Reported by Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="913a39927f1c5dcddd9e20befa1bb01ee9bb697a9dd69b7e6263006a6fdfef92" name="asterisk" arch="src">
  <version epoch="0" ver="18.10.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c065a4f8bf8006239b127302089d95ba53d060db770e8dbe363bdb5c9d74cadd" name="asterisk" arch="src">
  <version epoch="0" ver="18.10.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="37e9f9c3f9bd630b7886a0b69b0709c3b885b774ee91ef11b1ce1937ca87272a" name="asterisk" arch="src">
  <version epoch="0" ver="18.11.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992000">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c3d439d370b4cbaddb9e4aedc27121fc7af37dddbab3439daa186c9dc1b49e98" name="asterisk" arch="src">
  <version epoch="0" ver="18.11.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="bec8f93b33fa037eaf8f02df1002a525a74b1552c8e81a4b1e81ebd4092005da" name="asterisk" arch="src">
  <version epoch="0" ver="18.11.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="93f2d3f939ee7fa26367cb323c8b666eccca74d73157a2616d436ccca55682b6" name="asterisk" arch="src">
  <version epoch="0" ver="18.11.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="87086b6afe10a7bd56c5186e5f3add70dc0547d2f8dbcefb2dd250e6e6b48bff" name="asterisk" arch="src">
  <version epoch="0" ver="18.11.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="60c7885a00fc3fb97e9693426374881253c6342187c719e10d2bd73d17e82d8a" name="asterisk" arch="src">
  <version epoch="0" ver="18.12.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a4d3cad7bf2d2703c33ec0bd29d19118df4d6aaa68f18ccca0701279c365a899" name="asterisk" arch="src">
  <version epoch="0" ver="18.12.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f4f7de6fcbbbc5d70cdafa25d0881cf6d014cb4ad27cdf3133afe8b3159a3481" name="asterisk" arch="src">
  <version epoch="0" ver="18.13.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.10.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.10.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="c21dcf85318fdf449a270d3e66042e34ffc3127c38c2be7249dbc69471607dfa" name="asterisk" arch="src">
  <version epoch="0" ver="18.14.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.10.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="ee88de7bf5f15ed29a9fc8ef8a8fcc7466d82cce16d9a9012cb7e7e9bf700200" name="asterisk" arch="src">
  <version epoch="0" ver="18.14.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="4e31335ad3e2fbeff065aeee9e6357cda00bccb3d26b4c54186f0878fad80dad" name="asterisk" arch="src">
  <version epoch="0" ver="18.14.0" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="44cf6577fd6d1d318f2ec3c3b51a37e117d76e08f1557a08879e934a3cdc2634" name="asterisk" arch="src">
  <version epoch="0" ver="18.15.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f0ecf586964f460ee90df09495ef570ff81f69ab6d48c3c5054ff15a59f8ab83" name="asterisk" arch="src">
  <version epoch="0" ver="18.15.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="d59cf041939d49fe8d5deda1b463ac87d3401a6729ce88c15c752eba7c1a6e72" name="asterisk" arch="src">
  <version epoch="0" ver="18.16.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.11.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.11.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.11.3 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.11.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a98faa3a294ba81be30bad917810131a8f9b927c2e5beb6d3738820ee02ebf23" name="asterisk" arch="src">
  <version epoch="0" ver="18.17.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f00d61be61418ab161500ff25a2157aa6ffa0dde3f698c7ffe31456396eb1537" name="asterisk" arch="src">
  <version epoch="0" ver="18.17.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.12.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.12.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.12.1 resolves an issue reported by the community
- and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.12.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b67050551aaf89d28f02a2d1a3af09350c9a18186d6f73aacc855754b30f0420" name="asterisk" arch="src">
  <version epoch="0" ver="18.18.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.13.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.13.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.13.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.13.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
</package>
<package pkgid="3c023f3c9faf64ca2915a10daf3ff553848538b538fcbaa65a8007eb2f20b069" name="asterisk" arch="src">
  <version epoch="0" ver="18.18.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.14.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.14.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.14.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="00dd3a2d8e4b0fa65fba6ae251450fa44c92fbdf46874bba23d85ca4ec37c628" name="asterisk" arch="src">
  <version epoch="0" ver="18.19.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="eb504901109f2073b1367b98025198280add7796a4b858c36b04bd4a67434bc2" name="asterisk" arch="src">
  <version epoch="0" ver="18.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.0-1" date="1591876800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing (Reported
-                    by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false warning
-                    (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting call
-                    (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system (Reported
-                    by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP clients
-                    crash or not render received audio (Reported by Peter
-                    Turczak)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="3972ef2151e835a73b5af5fd736fbacb4ee42b35067281bacaea3392a46de6f0" name="asterisk" arch="src">
  <version epoch="0" ver="18.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.5.1-1" date="1592308800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.5.1. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.5.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.5.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="744a2194f8cacc7eb08dd10d185039415cb94600efdbd120436d3b9e32fc1b52" name="asterisk" arch="src">
  <version epoch="0" ver="18.2.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.6.0-1" date="1594900800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.6.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel (Reported
-                    by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by Kevin
-                    Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1dca27c1cc0622e0cfea41f437fde42e150cb201a97d97c1a8ca21324eb8583c" name="asterisk" arch="src">
  <version epoch="0" ver="18.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.14.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion. 
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="d9bb7af3afc44612b8acd355ff33d1de73338d824983bb37dabf9b1b7cf70a2f" name="asterisk" arch="src">
  <version epoch="0" ver="18.20.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.15.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.15.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="58563fb30a451838a6e67351fda094bcade012ec97aedf13c99f2e8a295d5016" name="asterisk" arch="src">
  <version epoch="0" ver="18.20.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.15.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="78746995ab6a41b6151e27cb599418dbab0e183a2a02f120108eda5c052eb055" name="asterisk" arch="src">
  <version epoch="0" ver="18.21.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.16.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.16.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.16.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.16.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="669a87f1aa8b753f02736ab883d1f1f15a64a3417fbc489fbadbd847623a11d0" name="asterisk" arch="src">
  <version epoch="0" ver="18.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="a61d817561ebf041a41fcd3517a0deab4ba8e0c7b664dad6cea434417308dfd2" name="asterisk" arch="src">
  <version epoch="0" ver="18.23.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.17.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.17.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.17.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.17.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="0dc95458485edec226aa27df74073e255874f08c4507e4e608670eeee2ada7d2" name="asterisk" arch="src">
  <version epoch="0" ver="18.23.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.18.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 18.18.0
- ==============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#40)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines. (#36)
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- pbx_dundi: Add PJSIP support.
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- res_calendar: output busy state as part of show calendar.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- app_queue: periodic announcement configurable start time.
- func_json: Fix JSON parsing issues.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- make_version: Strip svn stuff and suppress ref HEAD errors
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- res_agi: RECORD FILE plays 2 beeps.
- app_senddtmf: Add SendFlash AMI action.
- contrib: rc.archlinux.asterisk uses invalid redirect.
- main/iostream.c: fix build with libressl
- res_http_media_cache: Introduce options and customize
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines. (#36)
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="6504b7054940c189ef4d9dfe91aef945f29fce0dc909c5bab75442e7befb985a" name="asterisk" arch="src">
  <version epoch="0" ver="18.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.18.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.18.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.18.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 18.18.1
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.18.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.0...18.18.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.18.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending
-   a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute. This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="fdf9b39c9b6c6e9afc655e02924b703b4216369b4128e9f8f11311d582147e97" name="asterisk" arch="src">
  <version epoch="0" ver="18.24.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.19.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.19.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.19.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 18.19.0
- ==============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.19.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.18.1...18.19.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.19.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#76)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This was
-     fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="a9d07145d6ebc78b23aff0043d84d7531f6bd22193392e2133528e8d464f1bab" name="asterisk" arch="src">
  <version epoch="0" ver="18.24.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.0
- =======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.19.0...18.20.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added VoicemailBoxSummary -
-     Generate message list for a given mailbox VoicemailRemove - Remove a message
-     from a mailbox folder VoicemailMove - Move a message from one folder to
-     another within a mailbox VoicemailForward - Copy a message from one folder in
-     one mailbox to another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to
-     mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="31b1fe20c346f8814235e6ce491c722cbf2f2638072d74e191df3bc7ad95541b" name="asterisk" arch="src">
  <version epoch="0" ver="18.24.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.20.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-  initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-18.20.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.0...18.20.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="a55d2937a67ddf5273e5a724d82b90a2352550c2a1024e76d25963e4fa28f188" name="asterisk" arch="src">
  <version epoch="0" ver="18.25.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.20.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.20.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.20.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.20.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.20.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.1...18.20.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.20.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="2c7c0d11e429b9d862673ca7abfb7a89ea8a692e1dc4ed05cd7cdb0e2ef612f2" name="asterisk" arch="src">
  <version epoch="0" ver="18.26.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.21.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.21.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.21.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release asterisk-18.21.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.21.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.20.2...18.21.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.21.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al.: Allow adding all cflags to buildopts.h
-                                The "Build Options" entry in the "core show
-                                settings" CLI command has been renamed to "ABI
-                                related Build Options" and a new entry named
-                                "All Build Options" has been added that shows
-                                both breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-       before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-       EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-       in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-       Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-       nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-       provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-       iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-       cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-       topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-       around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-       output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-       built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-       multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
</package>
<package pkgid="a3101a81a6adc4bc4474175f05f25683c2c8684f0cbfd20cb10ee5e1419949e6" name="asterisk" arch="src">
  <version epoch="0" ver="18.26.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.22.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-18.22.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.22.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.22.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.22.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.22.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.21.0...18.22.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.22.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues.
-     See https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the existing
-     objects needed to be changed or removed, and new ones added. Additionally,
-     if res_stir_shaken is enabled in menuselect, you'll need to either have the
-     development package for libjwt v1.15.3 installed or use the
-     --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if available
-                      on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to move
-                      on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
- 
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application                                     
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.                                           
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process                               
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.                                           
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="861069cace4c012397fce49205f278f78b837dd5b1e43d83be099686cbb195e8" name="asterisk" arch="src">
  <version epoch="0" ver="18.26.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.23.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.23.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.22.0...18.23.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.23.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.23.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.23.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.23.1
-
- Change Log for Release asterisk-18.23.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.23.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.0...18.23.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.23.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-   res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.23.1...18.24.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 19
- Commit Authors: 8
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-18.24.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.24.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.0...18.24.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-      - voicemail.conf.sample: Fix ':' comment typo
-      - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.2
-
- Change Log for Release asterisk-18.24.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.1...18.24.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                         Write=originate, is sufficient permissions for code
-                         execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-  - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.24.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 18.24.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.24.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.24.3
-
- Change Log for Release asterisk-18.24.3
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.24.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.2...18.24.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.24.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-      - res_resolver_unbound: Test for NULL ub_result in
-        unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.25.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.25.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.25.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.25.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.25.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.25.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.24.3...18.25.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.25.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-18.26.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-18.26.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 7
- Issues Resolved: 13
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/`. This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/`. Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length
-            of up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the
-            `DB*` dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- George Joseph: (10)
- Jiangxc: (1)
- Naveen Albert: (6)
- Peter Jannesen: (2)
- Sean Bright: (7)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.1
-
- Change Log for Release asterisk-18.26.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.0...18.26.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.26.2-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 18.26.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/18.26.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 18.26.2
-
- Change Log for Release asterisk-18.26.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.26.1...18.26.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-  - [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-  - [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ------------------------
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-  - res_pjsip_messaging.c: Mask control characters in received From display name
-  - asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
-  asterisk.c: Add option to restrict shell access from remote consoles.
-  res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author: George Joseph
-             Date:   2025-05-19
-
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-             has been added that, when set, will prevent remote consoles from
-             executing shell commands using the '!' prefix.
-
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author: George Joseph
-                        Date:   2025-03-24
-
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="36a7d88f8eeb5950f92d71afe0796e98de1c4a9a7442c49fca02fa6c2d4befc3" name="asterisk" arch="src">
  <version epoch="0" ver="18.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.7.0-1" date="1599220800">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.7.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported by
-                    Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="981569b172698359c1328461f2e9ee666ffb77173889981dd10e94feaa6cef5a" name="asterisk" arch="src">
  <version epoch="0" ver="18.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 17.8.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 17.8.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 17.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it is
-                    not sent by email/processed by the mailcmd command (Reported
-                    by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on failed
-                    re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string to
-                    each message block. (Reported by Thomas Johnson)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-17.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195201">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="a95c750d86dc53acc137e0e359129ee43d255c94b662f0f0ec23400aaa89933f" name="asterisk" arch="src">
  <version epoch="0" ver="18.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.0-1" date="1603195200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.0.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-28589] - chan_sip: Depending on configuration an INVITE can alter
-                    Addr of a peer (Reported by Andrey V. T.)
- [ASTERISK-28580] - Bypass SYSTEM write permission in manager action allows
-                    system commands execution (Reported by Eliel Sardañons)
- [ASTERISK-28495] - res_pjsip_t38: 200 OK with SDP answer with declined stream
-                    causes crash (Reported by Alexei Gradinari)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-6863] - [patch] allow Asterisk to set high ToS bits as non-root on
-                   Linux (Reported by Matt Addison)
- [ASTERISK-17491] - CURLOPT() needs a "followlocation" parameter / "maxredirs"
-                    doesn't do anything (Reported by candrews)
- [ASTERISK-28639] - res_pjsip_endpoint_identifier_ip: Add ability to match on
-                    source port (Reported by Sean Bright)
- [ASTERISK-28614] - app_senddtmf: Allow "receiving" DTMF with PlayDTMF instead
-                    of only "sending" (Reported by lvl)
- [ASTERISK-28613] - func_curl: CURLOPT cannot set Content-Type header (Reported
-                    by Martin Tomec)
- [ASTERISK-28533] - func_jitterbuffer: Add support for video synchronization
-                    (Reported by Joshua C. Colp)
- [ASTERISK-17808] - [patch] Unregister a realtime moh class (Reported by Byron
-                    Clark)
- [ASTERISK-28489] - Channel variable SIPFROMDOMAIN for chan_pjsip to setup From
-                    header URI domain (Reported by Stas Kobzar)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
- [ASTERISK-28975] - res_http_websocket: Text payload data doesn't necessary
-                    include trailing zero (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28951] - Inconsistent behaviour queues.conf when there is (not) a
-                    [general] section (Reported by Walter Doekes)
- [ASTERISK-28965] - res_pjsip: Apply outbound proxy to static contacts on AOR
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28930] - ./configure --without-ssl build failure (Reported by Jaco
-                    Kroon)
- [ASTERISK-28957] - chan_sip: chan_sip does not process 400 response to an
-                    INVITE. (Reported by Frederic LE FOLL)
- [ASTERISK-28886] - chan_pjsip: PJSIP_SC_NULL does not exist in pjproject 2.7.2
-                    (Reported by Jared Smith)
- [ASTERISK-28888] - res_corosync: causes asterisk crash in huge distributed
-                    environment. (Reported by Università di Bologna - CESIA
-                    VoIP)
- [ASTERISK-28954] - StreamEcho() only returns 1 active stream (Reported by Bill
-                    Kervaski)
- [ASTERISK-28955] - "setvar" doesn't work properly in dahdi-channels.conf
-                    (Reported by Marin Odrljin)
- [ASTERISK-28953] - res_pjsip_session: Preserve stream label (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28942] - res_sorcery_memory_cache: Individual object expiration
-                    behaves unexpectedly with full backend caching (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28950] - Stale code in app_queue to check untouched channel
-                    (Reported by Walter Doekes)
- [ASTERISK-28644] - Stale comment in app_queue about ring_entry exception
-                    (Reported by Walter Doekes)
- [ASTERISK-28952] - Queue wrapuptime sometimes not respected (based on stale
-                    lastcall time) (Reported by Walter Doekes)
- [ASTERISK-28938] - core_unreal / core_local: Add support for multistream and
-                    re-negotiation (Reported by Joshua C. Colp)
- [ASTERISK-28948] - ARI channel create doesn't referencing the channel_id
-                    parameter (Reported by sungtae kim)
- [ASTERISK-28939] - res_rtp_asterisk: Don't have send/receive buffers on
-                    non-WebRTC (Reported by Joshua C. Colp)
- [ASTERISK-28944] - bridge_softmix: Transitioning a stream from inactive -&gt;
-                    sendrecv/sendonly doesn't re-negotiation (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28923] - T.38 Segfaults in chan_pjsip_queryoption (Reported by Yury
-                    Kirsanov)
- [ASTERISK-28940] - /channels/create doesn't get any parameters from the body
-                    (Reported by sungtae kim)
- [ASTERISK-28936] - res_pjsip: crash when dialing non-sip uri (Reported by
-                    Walter Doekes)
- [ASTERISK-28900] - res_fax: Double frame free when gateway in use with
-                    off-nominal format usage (Reported by Gregory Massel)
- [ASTERISK-28929] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28932] - res_pjsip_logger writing too big packets (Reported by
-                    nappsoft)
- [ASTERISK-28920] - bridge show all causes crash (Reported by sungtae
-                    kim)
- [ASTERISK-28921] - Wrong return value check for fwrite when writing to pcap
-                    file (Reported by nappsoft)
- [ASTERISK-28794] - res_pjsip: Crash when escaping during URI printing
-                    (Reported by nappsoft)
- [ASTERISK-28884] - x-ast-orig-host not filtered out from request URI and To
-                    header (Reported by nappsoft)
- [ASTERISK-28871] - res_pjsip_session: Unnecessary re-Invite on call answer
-                    (Reported by Alexei Gradinari)
- [ASTERISK-28903] - res_srtp: Answered Crypto Suite might be wrong in SDP/SDES.
-                    (Reported by Alexander Traud)
- [ASTERISK-28898] - bridge_softmix: Conference bridge not passing silent rtp
-                    packets (Reported by Jonathan Hunter)
- [ASTERISK-28892] - res_musiconhold: Module res_musiconhold throws false
-                    warning (Reported by Nicholas John Koch)
- [ASTERISK-28904] - RTP ICE leaks the memory (Reported by sungtae
-                    kim)
- [ASTERISK-26780] - res_pjsip: PJSIP Registration Fails when
-                    transport=transport-udp6 (Reported by Peter Sokolov)
- [ASTERISK-28854] - SIGSEGV when pjsip show history encounters IPV6 address
-                    (Reported by Roger James)
- [ASTERISK-28797] - [patch] tcptls: Fix notice when TLS is enabled but not
-                    configured. (Reported by Alexander Traud)
- [ASTERISK-28804] - [patch] app_osplookup.c: Avoid a format truncation.
-                    (Reported by Alexander Traud)
- [ASTERISK-28776] - Non async-signal-safe syscalls used after fork before exec
-                    (Reported by nappsoft)
- [ASTERISK-28870] - streams: One memory leak and one issue cloning streams
-                    (Reported by George Joseph)
- [ASTERISK-28829] - app_queue: leaking stasis subscription when Redirecting
-                    call (Reported by lvl)
- [ASTERISK-25844] - app_queue: Ghost channels in "core show channels" output
-                    (Reported by Etienne Lessard)
- [ASTERISK-28859] - pjsip: Increase maximum candidate count (Reported by Joshua
-                    C. Colp)
- [ASTERISK-22920] - Crash while Forwarding from TLS extension with CHANNEL args
-                    secure_bridge_media and secure_bridge_signaling (Reported by
-                    Shlomi Gutman)
- [ASTERISK-28852] - Unprotected access to nochecksums variable, causes build
-                    failures (Reported by Guido Falsi)
- [ASTERISK-28848] - app_fax: Compile. (Reported by Alexander
-                    Traud)
- [ASTERISK-28846] - stream: Enforce formats immutability (Reported by Joshua C.
-                    Colp)
- [ASTERISK-28847] - ARI channels cuts the endpoint string over 80 characters
-                    (Reported by sungtae kim)
- [ASTERISK-28811] - Crash occurs when fax session switches from T.38 to audio
-                    (Reported by Alexey Vasilyev)
- [ASTERISK-28839] - Sporadic crashes with Segmentation fault (Reported by
-                    Joeran Vinzens)
- [ASTERISK-28835] - IPv6 addresses in SDP incorrectly formatted (Reported by
-                    Daniel Heckl)
- [ASTERISK-28372] - Asterisk REPLY Wrong Contact header port (TCP) (Reported by
-                    Anton Satskiy)
- [ASTERISK-24428] - Document that Asterisk will use the default SIP ports (5060
-                    for TCP, 5061 for TLS) if the extern option variants aren't
-                    used (Reported by sstream)
- [ASTERISK-28838] - AST_MODULE_INFO requires, MODULEINFO does not mention
-                    (Reported by Alexander Traud)
- [ASTERISK-28841] - app_confbridge: Add support for disabling text messaging
-                    for a user (Reported by Joshua C. Colp)
- [ASTERISK-28837] - pjproject_bundled: Honor --without-pjproject. (Reported by
-                    Alexander Traud)
- [ASTERISK-28827] - res_rtp_asterisk: Loop when receive buffer is flushed by a
-                    received packet that is also in receive buffer with NACK
-                    (Reported by nappsoft)
- [ASTERISK-27195] - chan_sip: only sets ToS bits on UDP socket, ignoring TCP
-                    and TLS sockets (Reported by Joshua Roys)
- [ASTERISK-28826] - res_rtp_asterisk: Duplicate seqnos being added to send
-                    buffer with NACK (Reported by nappsoft)
- [ASTERISK-28812] - First DTMF is not get (Reported by Bernard
-                    Merindol)
- [ASTERISK-28758] - pjsip startup errors when using "with-ssl" configure option
-                    (Reported by Patrick Wakano)
- [ASTERISK-28824] - BuildSystem: Search for Python/C API when possibly needed
-                    only. (Reported by Alexander Traud)
- [ASTERISK-27717] - [patch] BuildSystem: In NetBSD, the Python Programming
-                    Language is python-2.7. (Reported by Alexander Traud)
- [ASTERISK-28817] - chan_pjsip: constant DTMF tone if RTP is not setup yet
-                    (Reported by Kevin Harwell)
- [ASTERISK-28819] - [patch] bridge_softmix_binaural: Show state in menuselect.
-                    (Reported by Alexander Traud)
- [ASTERISK-28816] - [patch] BuildSystem: Remove doc/tex and doc/pdf leftovers.
-                    (Reported by Alexander Traud)
- [ASTERISK-28818] - [patch] BuildSystem: Allow space in path. (Reported by
-                    Alexander Traud)
- [ASTERISK-28809] - [patch] res_rtp_asterisk: Avoid absolute value on unsigned
-                    subtraction. (Reported by Alexander Traud)
- [ASTERISK-28796] - func_channel: cannot read fields exten, context, userfield,
-                    channame from dialplan (Reported by Sébastien Duthil)
- [ASTERISK-28803] - [patch] chan_unistim: Avoid tautological warnings with
-                    clang. (Reported by Alexander Traud)
- [ASTERISK-28808] - [patch] test_stasis: Avoid always true warning with clang.
-                    (Reported by Alexander Traud)
- [ASTERISK-28056] - res_pjsip: Incorrect endpoint status after endpoint
-                    synchronization for a specific AOR (Reported by Jason Hord)
- [ASTERISK-28795] - channel: write to a stream on multi-frame writes (Reported
-                    by Kevin Harwell)
- [ASTERISK-28789] - test_utils: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28788] - func_aes: incorrectly printing error 'declined to load'
-                    (Reported by Alexander Traud)
- [ASTERISK-28790] - Crash during conference call using confbridge and video
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-16676] - DAHDIRAS fails to properly initiate pppd unless asterisk is
-                    running as root (Reported by Jaco Kroon)
- [ASTERISK-21205] - [patch] dundi_read_result crash due to negative number
-                    (Reported by Jaco Kroon)
- [ASTERISK-28784] - res_pjsip_sdp_rtp: Only do hold/unhold on first audio
-                    stream (Reported by Joshua C. Colp)
- [ASTERISK-28743] - Asterisk is crashing if the 200 OK with SDP (Reported by
-                    sungtae kim)
- [ASTERISK-28783] - res_pjsip_session: Allow default non-audio streams to have
-                    reflected state (Reported by Joshua C. Colp)
- [ASTERISK-28774] - chan_pjsip's rtptimeout is erroneously triggered during
-                    direct-media (native_rtp) bridge (Reported by Michael
-                    Neuhauser)
- [ASTERISK-20325] - Comments in configs/func_odbc.conf.sample are not
-                    consistent with examples. Missing examples. (Reported by
-                    Olivier Krief)
- [ASTERISK-28780] - app_mixmonitor: Memory leak due to race condition between
-                    AMI MixMonitor and hangup (Reported by Joshua C. Colp)
- [ASTERISK-28773] - Incorrect Sender SSRC in RTCP when p2p rtp bridge is active
-                    (Reported by Torrey Searle)
- [ASTERISK-28769] - DTLS Handshake Fails to Occur if ice_support is enabled but
-                    not used (Reported by Torrey Searle)
- [ASTERISK-28759] - A non negotiated rtp frame causes call disconnection when
-                    there is a SSRC change (Reported by Paulo Vicentini)
- [ASTERISK-26711] - func_enum: ENUM code wrong case (Reported by
-                    Vitold)
- [ASTERISK-23407] - Fix the FSF address in the headers of lots of pjproject
-                    files (Reported by Jared Smith)
- [ASTERISK-19460] - [patch] Function TXTCIDNAME never actually makes DNS calls
-                    and always returns an empty string (Reported by George
-                    Joseph)
- [ASTERISK-28766] - PJSIP blind transfer not completed after using Proceeding()
-                    (Reported by lvl)
- [ASTERISK-28764] - res_rtp_asterisk: Improve NACK support and seqno handling
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28755] - SIP/Stasis: SIP headers not transmitted in the "variables"
-                    field (Reported by Jean Aunis - Prescom)
- [ASTERISK-28685] - check_expr2: linking (when hardening) and cross-compiling
-                    troubles (Reported by Sebastian Kemper)
- [ASTERISK-28754] - ASTERISK-28738 Causes Audio Issue After Hold (Reported by
-                    Ross Beer)
- [ASTERISK-28697] - res_pjsip: Named ACL does not update on reload if changed
-                    (Reported by Timothy Vanderaerden)
- [ASTERISK-28746] - res_pjsip_outbound_registration keeps retrying the first
-                    entry in a SRV record set (Reported by George Joseph)
- [ASTERISK-28716] - ICE: pjnath shouldn't wait for ICE to complete before
-                    allowing sending (Reported by Benjamin Keith Ford)
- [ASTERISK-28738] - Incorrect state machine used when MOH_PASSTHRU is used
-                    (Reported by Torrey Searle)
- [ASTERISK-28742] - res_rtp_asterisk: static for audio due to incomplete
-                    dtls/srtp setup (Reported by Kevin Harwell)
- [ASTERISK-28735] - Realtime MoH Unknown format '' -- defaulting to SLIN
-                    (Reported by Ross Beer)
- [ASTERISK-28730] - res_pjsip_session: Fix out of order session refreshes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-26955] - pjsip: SIP Packets with Via "received=" Containing IPv6
-                    Address Delimited by "[]" Rejected (Reported by Peter
-                    Sokolov)
- [ASTERISK-28718] - chan_sip: Returns 403 if RTP ports are depleted, should
-                    return 503 (Reported by Walter Doekes)
- [ASTERISK-28713] - res_stasis_playback: Error building JSON (Reported by
-                    Sébastien Duthil)
- [ASTERISK-28714] - REGRESSION: Feature subscription_persistence_recreate
-                    (ASTERISK-27759) Causes Segfaults (Reported by Ross Beer)
- [ASTERISK-26082] - res_pjsip_messaging: MessageSend Content-Type can't be
-                    changed (Reported by Alex)
- [ASTERISK-28423] - ARI causes STASIS Deadlock (Reported by Ross
-                    Beer)
- [ASTERISK-28679] - stasis application is destroyed after its creation
-                    (Reported by Francois Blackburn)
- [ASTERISK-25421] - PJSIP. MESSAGE_SEND_STATUS set to SUCCESS in spite of the
-                    error when sending (Reported by Dmitriy Serov)
- [ASTERISK-28686] - chan_sip strictrtp=yes fails when media source is changed:
-                    no audio (Reported by Walter Doekes)
- [ASTERISK-28139] - RTP Stream Incorrect Payload Type Causes Asterisk To Drop
-                    Calls (Reported by Paul Brooks)
- [ASTERISK-28677] - CDR billsec is always 0 for transferred calls (Reported by
-                    Maciej Michno)
- [ASTERISK-28702] - chan_dahdi: holding a channel via flash to dialtone times
-                    out after 0:16:40 (Reported by Andrew Siplas)
- [ASTERISK-24484] - Update documentation for statsd module - usage requirements
-                    unclear (Reported by Dan Jenkins)
- [ASTERISK-28706] - silk 24hHz doesn't show up in 'core show translation'
-                    output (Reported by Sean Bright)
- [ASTERISK-28695] - core: minmemfree watermark uses free RAM, not available RAM
-                    (Reported by Kevin Flyn)
- [ASTERISK-28693] - chan_sip: SIP MESSAGE beginning with a whitespace appears
-                    empty in the dialplan (Reported by Frank Matano)
- [ASTERISK-23739] - [patch]Segfault forwarding voicemail with ODBC storage
-                    enabled and realtime voicemail_data is used (Reported by Stas
-                    Kobzar)
- [ASTERISK-27622] - empty voicemail.conf required for ARA (realtime) voicemail
-                    to leave message (Reported by Jim Van Meggelen)
- [ASTERISK-21794] - CLI command 'realtime update2' syntax failure when using
-                    according to usage help (Reported by Cedric BASSAGET)
- [ASTERISK-28349] - Pause reason not reported in QueueMember AMI event
-                    (Reported by Niksa Baldun)
- [ASTERISK-25429] - res_pjsip_endpoint_identifier_ip: Document support for
-                    hostnames (Reported by Joshua C. Colp)
- [ASTERISK-27775] - res_pjsip_notify: Multiple Event headers can be present
-                    instead of just one (Reported by AvayaXAsterisk)
- [ASTERISK-28682] - app_record: Lack of `beep` audio file causes application to
-                    return error and hangup (Reported by Corey Farrell)
- [ASTERISK-28507] - Wiki docs missing for MessageWaiting (Reported by David M.
-                    Lee)
- [ASTERISK-27759] - res_pjsip_pubsub: Subscription persistence does not
-                    preserve XML version number (Reported by Bryan Nelson)
- [ASTERISK-28605] - chan_dahdi: Deadlock in Hangup Scenarios with concurrent
-                    command pri show span X (Reported by Dirk Wendland)
- [ASTERISK-28633] - stasis bridge topic leak (Reported by Joeran
-                    Vinzens)
- [ASTERISK-28492] - pjsip reload not reloading wizard endpoint/pickup_group
-                    endpoint/call_group (Reported by Jean-Denis Girard)
- [ASTERISK-28562] - SIP WSS message not processed until next frame arrives
-                    (Reported by Robert Sutton)
- [ASTERISK-28667] - Asterisk ignores parsing of config files if a Byte order
-                    mark is present (Reported by Robin Leffmann)
- [ASTERISK-28625] - Playback of local files impacted by large media cache
-                    (Reported by Kevin Reeves)
- [ASTERISK-27243] - contrib: valgrind.supp doesn't suppress what it's supposed
-                    to due to invalid syntax (Reported by Richard Kenner)
- [ASTERISK-28664] - "trustrpid" is misspelled in sip_to_pjsip.py (Reported by
-                    Pascal Cadotte Michaud)
- [ASTERISK-28636] - app_chanisavail+cdr: ChanIsAvail sometimes fails to
-                    deactivate CDR. (Reported by Frederic LE FOLL)
- [ASTERISK-28604] - app_meetme, chan_ooh323 and cdr_mysql don't build on 17.0.0
-                    (Reported by George Joseph)
- [ASTERISK-28659] - res_pjsip_sdp_rtp: Bundle includes non-existent media
-                    stream if codecs create additional streams and offer does not
-                    have them (Reported by nappsoft)
- [ASTERISK-28660] - res_fax: wrap Asterisk initiated negotiation with config
-                    option (Reported by Kevin Harwell)
- [ASTERISK-28626] - Missing arguments in PJSIP_CONTACT function documentation
-                    (Reported by Pascal Cadotte Michaud)
- [ASTERISK-28609] - Memory Leak in res_rtp_asterisk.c (Reported by Ted
-                    G)
- [ASTERISK-28651] - chan_sip logs errors on tx to non-existent TCP connections
-                    (Reported by Jaco Kroon)
- [ASTERISK-28502] - chan_pjsip incorrectly re-writes REGISTER 200 Response
-                    Contact (Reported by Ross Beer)
- [ASTERISK-28641] - res_pjsip Segfaults when realtime configuration to an AOR
-                    points to a not existent AOR (Reported by Ross Beer)
- [ASTERISK-28647] - chan_sip: RTP frames not transmitted after emitting a COLP
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28637] - chan_sip+native_bridge_rtp: directmedia compatibility check
-                    failure when negociated ptime is not default ptime. (Reported
-                    by Frederic LE FOLL)
- [ASTERISK-28445] - res_pjsip_session: ast_json_vpack: Invalid UTF-8 string on
-                    hangup when TEST_FRAMEWORK enabled (Reported by Bernhard
-                    Schmidt)
- [ASTERISK-28631] - res_parking: Doesn't park when parkee and parker are the
-                    same (Reported by Ross Beer)
- [ASTERISK-28621] - Enforce T.38 error correction mode at 200 ok received
-                    (Reported by Salah Ahmed)
- [ASTERISK-28624] - res_pjsip_outbound_registration: add SRV failover (Reported
-                    by Kevin Harwell)
- [ASTERISK-28608] - app_amd: Use time calculation to calculate timeout
-                    (Reported by Michael Cargile)
- [ASTERISK-28615] - chan_dahdi: PRI span status may stay "Down, Active" after a
-                    short alarm (Reported by Frederic LE FOLL)
- [ASTERISK-28576] - res_rtp_asterisk: ICE Completion Crash when sent packet
-                    length doesn't match (Reported by Joshua Elson)
- [ASTERISK-26481] - FILE function grabs garbage along with read data when
-                    target line has no newline (Reported by Jonathan Harris)
- [ASTERISK-28618] - bridge_softmix: hold not cleared when joining a softmix
-                    bridge (Reported by Kevin Harwell)
- [ASTERISK-28616] - parking: Deadlock when multi call parking (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28572] - Memory leaks in res_calendar_exchange and
-                    res_calendar_icalendar (Reported by Yoooooo Ha)
- [ASTERISK-28585] - ari/resource_events: Crash in event session cleanup
-                    (Reported by Kevin Harwell)
- [ASTERISK-28590] - utils.c throws repeated warnings;
-                    "pthread_attr_setstacksize: Invalid argument" (Reported by
-                    Speed Dial Dave)
- [ASTERISK-28578] - race condition on pjsip channelstats command (Reported by
-                    Salah Ahmed)
- [ASTERISK-28571] - cdr_pgsql: accesses obsolete (and finally removed) column
-                    (Reported by Christoph Moench-Tegeder)
- [ASTERISK-28575] - MWI Send Notify Crash on 16.6 (Reported by Joshua
-                    Elson)
- [ASTERISK-28574] - pjproject fails to build on 16.6.0, works on 16.5 (Reported
-                    by Niklas Larsson)
- [ASTERISK-28561] - Asterisk Deadlocks (Reported by
-                    Aheliotech)
- [ASTERISK-28086] - chan_pjsip: Crash when initiating PlayDTMF over AMI
-                    (Reported by Jeremiah Gadd)
- [ASTERISK-28552] - res_pjsip_mwi: Frack during unload on unsolicited_mwi
-                    container (Reported by Kevin Harwell)
- [ASTERISK-28566] - CDR backend unload problem during active call(s) (Reported
-                    by Marian Piater)
- [ASTERISK-28553] - stasis.c: Crash during unload (Reported by Kevin
-                    Harwell)
- [ASTERISK-28544] - Wrong contact representation in ipv6 mode (Reported by
-                    Jørgen H)
- [ASTERISK-28534] - Segmentation fault when there is no priority for an
-                    extension (Reported by Timothy Vanderaerden)
- [ASTERISK-28463] - res_pjsip_path: Crash when invalid contact is configured
-                    (Reported by Juan Martin)
- [ASTERISK-28521] - pjsip: Memory Leak (Reported by
-                    Mark)
- [ASTERISK-28523] - Asterisk 16.5.0 Memory leak (Reported by Cyril
-                    Ramière)
- [ASTERISK-28536] - Asterisk release candidates fail to build on FreeBSD
-                    (Reported by Guido Falsi)
- [ASTERISK-28538] - chan_pjsip: Deadlock on fax detection (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28497] - func_odbc: truncating Unicode string on readsql (Reported
-                    by Boris P. Korzun)
- [ASTERISK-23756] - setvar directive when used in template and a child of said
-                    template, results in duplicate variable names (Reported by
-                    Michael Goryainov)
- [ASTERISK-28527] - ChanIsAvail() creates a CDR if unanswered=yes is set in
-                    cdr.conf (Reported by Frederic LE FOLL)
- [ASTERISK-28525] - chan_dahdi: set CHANNEL(hangupsource) when a PRI channel
-                    hangs up (Reported by Frederic LE FOLL)
- [ASTERISK-28511] - codec_resample: Bad sound quality when up sampling from
-                    SLIN16 to SLIN32 (Reported by Ruddy G)
- [ASTERISK-28499] - translate: Crash when frame does not have a "src" field set
-                    (Reported by Gregory Massel)
- [ASTERISK-25592] - chan_unistim: Clang Warning: variable sized type not at end
-                    of a struct (Reported by Alexander Traud)
- [ASTERISK-28488] - pjsip mwi: n+1 sip notify's sent on re-register (Reported
-                    by Chris Savinovich)
- [ASTERISK-28509] - PJSIP cnonce generated on Linux contains 36 characters, NEC
-                    only supports up to 32 characters (Reported by Dan Cropp)
- [ASTERISK-28505] - app_voicemail/IMAP: segfault in leave_voicemail because not
-                    checking mailstream (Reported by Alexei Gradinari)
- [ASTERISK-28487] - compile menuselect on gentoo (Reported by
-                    Kilburn)
- [ASTERISK-28472] - Asterisk occasionally passes a NULL as srtp-&gt;session to
-                    srtp_protect/unprotect causing SEGV (Reported by Jonas
-                    Swiatek)
- [ASTERISK-28498] - cel / cdr: Event times may be incorrect (Reported by Joshua
-                    C. Colp)
- [ASTERISK-28480] - json integer overflow in ssrc and timestamp (Reported by
-                    Salah Ahmed)
- [ASTERISK-28228] - res_pjsip: pjsip show contacts prints double entries
-                    (Reported by Ian Jones)
- [ASTERISK-28483] - packet lost on UDPTL wrap around (Reported by Torrey
-                    Searle)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28959] - res_pjsip: Added option for disable rport parameter set
-                    (Reported by sungtae kim)
- [ASTERISK-28958] - Continue reading string when ping received by websocket
-                    (Reported by Nickolay V. Shmyrev)
- [ASTERISK-28945] - AMI SendText - add Content-Type parameter (Reported by
-                    Kevin Harwell)
- [ASTERISK-28949] - res_http_websocket: Add masking to websocket client
-                    (Reported by Moises Silva)
- [ASTERISK-28899] - Upgrade Asterisk to bundled pjproject 2.10 (Reported by
-                    Kevin Harwell)
- [ASTERISK-28895] - res_pjsip_logger: Add tons'o'functionality (Reported by
-                    Joshua C. Colp)
- [ASTERISK-28896] - ari: Add support for specifying variables on channel create
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28879] - pjproject has race conditions in it's build system
-                    (Reported by Guido Falsi)
- [ASTERISK-28866] - third-party/pjproject/configure.m4 contains bashisms
-                    (Reported by Guido Falsi)
- [ASTERISK-28853] - Missing include on FreeBSD (Reported by Guido
-                    Falsi)
- [ASTERISK-28832] - chan_mobile creates PCMA streams that make some VoIP
-                    clients crash or not render received audio (Reported by Peter
-                    Turczak)
- [ASTERISK-28813] - func_volume: Allow decimal numbers as parameter to improve
-                    granularity (Reported by Jean Aunis - Prescom)
- [ASTERISK-28777] - Codec Negotiation: add outgoing_call_offer_prefs option
-                    (Reported by Kevin Harwell)
- [ASTERISK-27946] - dial (API): Storage of dialed target uses AST_MAX_EXTENSION
-                    when it shouldn't (Reported by Joshua Elson)
- [ASTERISK-28782] - Add support for Content-Disposition header in multi-part
-                    INVITES (Reported by Torrey Searle)
- [ASTERISK-28787] - res_pjsip_session: Decide more intelligently when to add
-                    video (Reported by Joshua C. Colp)
- [ASTERISK-28756] - Codec Negotiation: add incoming_call_offer_pref option
-                    (Reported by Kevin Harwell)
- [ASTERISK-28750] - TLS/SSL Key too small error (Reported by Martin
-                    Zeh)
- [ASTERISK-28733] - stream: Add support for adding/removing streams during
-                    SFU/calls (Reported by Joshua C. Colp)
- [ASTERISK-24798] - Documentation - Clarify That Format Is Set By File Name
-                    Extension In MixMonitor (Reported by xrobau)
- [ASTERISK-28726] - install_prereq script uses the interactive mode when
-                    installing aptitude (Reported by Sylvain Afchain)
- [ASTERISK-28710] - Should be able to disable the /httpstatus URI in the
-                    built-in HTTP server (Reported by Sean Bright)
- [ASTERISK-28484] - Add AudioSocket support (Reported by Seán C.
-                    McCord)
- [ASTERISK-28638] - Simplify dialplan for Dial, Page, and ChanIsAvail (Reported
-                    by cmaj)
- [ASTERISK-28673] - GET FULL VARIABLE documentation clarification (Reported by
-                    Jonathan Harris)
- [ASTERISK-28629] - [patch] Add an "inhibitCOLP" flag to the bridges REST API
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-28658] - app_confbridge: Add support for setting maximum sample rate
-                    (Reported by Joshua C. Colp)
- [ASTERISK-28602] - res_pjsip_outbound_registration: Maximum retries reached
-                    (Reported by Daniel)
- [ASTERISK-28586] - Typo in README-SERIOUSLY.bestpractices.md (Reported by Sam
-                    Banks)
- [ASTERISK-22192] - [patch] Allow voicemail forwards with ODBC backend when
-                    format differs from attachfmt column (Reported by cmaj)
- [ASTERISK-28567] - Problem with ASTERISK-20207: Asterisk should clear out any
-                    .lock files in the voice mail directory on startup. (Reported
-                    by Michael)
- [ASTERISK-28542] - [patch] add the ability for asterisk to generate on-hold
-                    re-invites (Reported by Torrey Searle)
- [ASTERISK-28512] - Add pass-through support for H.265 (HEVC) codec (Reported
-                    by Florian Floimair)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9cc6043477dbe7facc54257ac2c20cec5a67313cda23e15725b63ebaa7670925" name="asterisk" arch="src">
  <version epoch="0" ver="18.5.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.0.1-1" date="1604577600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.37.1, 16.14.1, 17.8.1, 18.0.1 and 16.8-cert5.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
-     AST-2020-001: Remote crash in res_pjsip_session
-     Upon receiving a new SIP Invite, Asterisk did not return the created dialog
-     locked or referenced.
-
-     AST-2020-002: Outbound INVITE loop on challenge with different nonce.
-     If Asterisk is challenged on an outbound INVITE and the nonce is changed in
-     each response, Asterisk will continually send INVITEs in a loop. This causes
-     Asterisk to consume more and more memory since the transaction will never
-     terminate (even if the call is hung up), ultimately leading to a restart or
-     shutdown of Asterisk. Outbound authentication must be configured on the
-     endpoint for this to occur.
-
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.37.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.14.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.8.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.0.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert5
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-002.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9fa7bf826787aa809d6f9ac7a31a2240dae2bb796c4fa8bb0c15485f3d34a734" name="asterisk" arch="src">
  <version epoch="0" ver="18.5.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.0-1" date="1605873600">- The Asterisk Development Team would like to announce the release of Asterisk 18.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross
-                    Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
</package>
<package pkgid="8617b64512301eaccc2dc4f0655f7a901363d1cc9be1bb283051a05d9db72b72" name="asterisk" arch="src">
  <version epoch="0" ver="18.5.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.1.1-1" date="1608638400">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18. The available releases are released as versions
- 13.38.1, 16.15.1, 17.9.1 and 18.1.1.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2020-003: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP message is received that has a History-Info header,
-               which contains a tel-uri.
-
- AST-2020-004: Remote crash in res_pjsip_diversion A crash can occur in Asterisk
-               when a SIP 181 response is received that has a Diversion header,
-               which contains a tel-uri.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.1.1
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2020-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2020-004.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
</package>
<package pkgid="31557b6c8c9f267f0009c718dc0a8046005014f7f9a5445017a3427be6cb6c41" name="asterisk" arch="src">
  <version epoch="0" ver="18.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.0-1" date="1611230400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.2.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains
-                    History-Info
-                    (Reported by Torrey Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N
-                    GM)
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant
-                    Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="75c7e60b8fc8cb75394aea6aaf481dc892185aa265b40a7043f012121b04f0f1" name="asterisk" arch="src">
  <version epoch="0" ver="18.6.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.1-1" date="1613736000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available releases
- are released as versions 13.38.2, 16.16.1, 17.9.2, 18.2.1 and 16.8-cert6.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-001: Remote crash in res_pjsip_diversion if a registered user is
-               tricked into dialing
-
- AST-2021-002: Remote crash possible when negotiating T.38
-
- AST-2021-003: Remote attacker could prematurely tear down SRTP calls an
-               unauthenticated remote attacker could replay SRTP packets
-               which could cause an Asterisk instance configured without strict
-               RTP validation to tear down calls prematurely.
-
- AST-2021-004: An unsuspecting user could crash Asterisk with multiple
-               hold/unhold requests due to a signedness comparison mismatch, an
-               authenticated WebRTC client could cause a stack overflow and
-               Asterisk crash by sending multiple hold/unhold requests in quick
-               succession.
-
- AST-2021-005: Remote Crash Vulnerability in PJSIP channel driver given a
-               scenario where an outgoing call is placed from Asterisk to a
-               remote SIP server it is possible for a crash to occur.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert6
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-003.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-005.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
</package>
<package pkgid="63fd57e5d9509dc538e307ade1e902fbe95eaf4f52f25f71f16476efb5018d54" name="asterisk" arch="src">
  <version epoch="0" ver="18.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.2.2-1" date="1614859200">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 17 and 18, and Certified Asterisk 16.8. The available releases are
- released as versions 16.16.2, 17.9.3, 18.2.2 and 16.8-cert7.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2021-006: Crash when negotiating T.38 with a zero port When Asterisk sends
-               a re-invite initiating T.38 faxing and the endpoint responds with
-               a m=image line and zero port, a crash will occur in Asterisk.
-               This is a reoccurrence of AST-2019-004.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.16.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.2.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert7
-
- The security advisory is available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6662240d28ccb433fcfabaaa1381aedb43893fab3cc9719b8d07c8a183e6925a" name="asterisk" arch="src">
  <version epoch="0" ver="18.7.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.3.0-1" date="1616673600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.3.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah )
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5a97f825477b9837a585c54145e82c710e74b3c3605cebc04f6466969bdb72da" name="asterisk" arch="src">
  <version epoch="0" ver="18.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.4.0-1" date="1620648000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="53908618b83b639ed5ddd5e2b299facc0f52ce7f3ebdf17ac4586e2167bc44b6" name="asterisk" arch="src">
  <version epoch="0" ver="18.9.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.9.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.9.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.9.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29764] - chan_misdn: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.9.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="00ac1b787436b2145736ca5ac1dff3c088b68220370c57e6983b093212954b96" name="asterisk" arch="src">
  <version epoch="0" ver="19.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.0-1" date="1624536000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.5.0. This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="2f09ef521aee61d28a7ff9b85411291d903839e9914a91a9c1a4d7ccc84f84a9" name="asterisk" arch="src">
  <version epoch="0" ver="19.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-1" date="1627128000">- The Asterisk Development Team would like to announce security releases for
- Asterisk 13, 16, 17 and 18, and Certified Asterisk 16.8. The available
- releases are released as versions 13.38.3, 16.19.1, 17.9.4, 18.5.1 and
- 16.8-cert10.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- * AST-2021-007: Remote Crash Vulnerability in PJSIP channel driver when
- Asterisk receives a re-INVITE without SDP after having sent a BYE request a
- crash will occur. This occurs due to the Asterisk channel no longer being
- present while code assumes it is.
- * AST-2021-008: Remote crash when using IAX2 channel driver If the IAX2
- channel driver receives a packet that contains an
- * AST-2021-009: pjproject/pjsip: crash when SSL socket destroyed during
- handshake depending on the timing, it’s possible for Asterisk to crash when
- using a TLS connection if the underlying socket parent/listener gets destroyed
- during the handshake.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-13.38.3
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.19.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-17.9.4
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.5.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert10
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2021-007.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-008.pdf
- https://downloads.asterisk.org/pub/security/AST-2021-009.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="619b3d691f447d21c796d3d94948ed7002afd77ea56ab685e73eecf696cc3431" name="asterisk" arch="src">
  <version epoch="0" ver="19.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-2" date="1627992000">- Added support level of extra modules of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992001">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="04d15c93de33439973ef500d2a2dc7d7aa47f8cb666450a5f1e0b4fa1afcbe29" name="asterisk" arch="src">
  <version epoch="0" ver="19.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.5.1-3" date="1627992000">- Added OGG/Speex patch to avoid crash when writting speex file formats</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="9226089fb3d3f9b67ae45ace5fcc7547cd25a3368ab7ea9a6617f7440386d3db" name="asterisk" arch="src">
  <version epoch="0" ver="19.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-1" date="1628856000">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="90b57a1501f5cdbd25c6e7869353c123cdfb60487854ff0e758f8ac7dab5948e" name="asterisk" arch="src">
  <version epoch="0" ver="19.3.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.6.0-2" date="1632052800">- Added app_audiofork subpackage
- Renamed {opus,vp8,amr,evs} packages to codec_{opus,vp8,amr,evs}
- Renamed json package to res_json</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="657fbea71cac44a135f085ca2243ecd5c41ef92613e4f1ae0b0fa5457687df99" name="asterisk" arch="src">
  <version epoch="0" ver="19.3.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.0-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29554] - cdr_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29555] - app_mysql: Deprecated in 1.8, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29557] - app_ices: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29559] - app_fax: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29560] - app_url: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29561] - app_image: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29562] - app_nbscat: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29563] - app_dahdiras: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29564] - cdr_syslog: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29565] - chan_oss: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29566] - chan_phone: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29568] - chan_nbs: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29569] - chan_misdn: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29570] - chan_vpb: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29571] - res_config_sqlite: Deprecated in 16, to be removed in 19
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29573] - conf2ael: Deprecated in 16, to be removed in 19 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29574] - muted: Deprecated in 16, to be removed in 19 (Reported by
-                    Joshua C. Colp)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953601">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
</package>
<package pkgid="a773c75f81acebf8872d7030d2bde6ad247a11bb2792591350002063913dbb33" name="asterisk" arch="src">
  <version epoch="0" ver="19.3.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.7.1-1" date="1633953600">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.7.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.7.1 resolves an issue reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29685] - pbx_ael: Infinite loop on reload (Reported by Joshua C.
-                    Colp)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.7.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="eaeec1a0c7396273cab505cabd70f57ec1ab1937409a0edce483d7f038f5caa9" name="asterisk" arch="src">
  <version epoch="0" ver="19.3.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 18.8.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 18.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 18.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854401">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b561d47cef091185070b3aeb21f79eed2dc1b64a224d8ffdd23088fdaadf4b02" name="asterisk" arch="src">
  <version epoch="0" ver="19.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.0.0-1" date="1635854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29305] - ASTERISK-29203 / AST-2021-002 -- Another scenario is
-                    causing a crash (Reported by Gregory Massel)
- [ASTERISK-29260] - sRTP Replay Protection ignored; even tears down long calls
-                    (Reported by Alexander Traud)
- [ASTERISK-29227] - res_pjsip_diversion: sending multiple 181 responses causes
-                    memory corruption and crash (Reported by Ivan Poddubny)
- [ASTERISK-29219] - res_pjsip_diversion: Crash if Tel URI contains History-Info
-                    (Reported by Torrey Searle)
- [ASTERISK-29057] - pjsip: Crash on call rejection during high load (Reported
-                    by Sandro Gauci)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-29454] - New application to reload modules (Reported by N A)
- [ASTERISK-29444] - Add application to wait for condition (Reported by N A)
- [ASTERISK-29442] - app_dial: Expand A option to allow announcement playback to
-                    caller (Reported by N A)
- [ASTERISK-29446] - app_confbridge: New ConfKick application (Reported by N A)
- [ASTERISK-29440] - app_confbridge: Allow ConfBridge answer to be suppressed
-                    (Reported by N A)
- [ASTERISK-29431] - Minimum and maximum dialplan functions (Reported by N A)
- [ASTERISK-29439] - func_volume: Volume function can't be read (Reported by N
-                    A)
- [ASTERISK-27477] - Chan_pjsip does not support unauthenticated OPTIONS ping
-                    (Reported by Ross Beer)
- [ASTERISK-29027] - Implement support for History-Info (Reported by Torrey
-                    Searle)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-20219] - [patch] - IAX2 Call Encryption Fails with RSA
-                    authentication (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don't work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
- [ASTERISK-27871] - Remote URL in playback must end with file extension
-                    (Reported by Caesar)
- [ASTERISK-29507] - STUN timeout is silently delaying calls (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29514] - ari: Audiosocket segfault when no data specified (Reported
-                    by Igor Goncharovsky)
- [ASTERISK-29503] - Updated identify/match syntax not supported by config
-                    wizard (Reported by Sean Bright)
- [ASTERISK-29480] - fixedjitterbuffer contains an un-wrappered assert that
-                    triggers on a negative time slew (Reported by Dan Cropp)
- [ASTERISK-29485] - core: Inband generation of tones for Busy() and
-                    Congestion() may not occur (Reported by Joshua C. Colp)
- [ASTERISK-29479] - [patch] Channels are not put on hold for Session Progress
-                    with inactive audio (Reported by Bernd Zobl)
- [ASTERISK-29475] - SayNumber triggers WARNING if caller hangs up during
-                    application execution (Reported by N A)
- [ASTERISK-29404] - Consolidate res_pjsip_messaging fixes for domain name
-                    (Reported by George Joseph)
- [ASTERISK-29441] - Core reload making TCP endpoints go offline (Reported by
-                    Luke Escude)
- [ASTERISK-28237] - "FRACK!, Failed assertion bad magic number" happens when
-                    unsubscribe an application from an event source (Reported by
-                    Lucas Tardioli Silveira)
- [ASTERISK-28393] - Multidomain support issue (Reported by Andrea Sannucci)
- [ASTERISK-29433] - res_rtp_asterisk: Server reflexive candidates use incorrect
-                    raddr for RTCP (Reported by Chris)
- [ASTERISK-29397] - pjsip: Asterisk isn't tolerant of RFC8760 UASs (Reported by
-                    George Joseph)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-29372] - file.c switch does not account for flash events (Reported
-                    by N A)
- [ASTERISK-29370] - chan_sip does not recognize application/hook-flash
-                    (Reported by N A)
- [ASTERISK-29377] - cpool_release_pool "double free or corruption (out)"
-                    (Reported by Robert Sutton)
- [ASTERISK-29358] - chan_pjsip: Trace message for progress is output even if
-                    frame is not queued (Reported by Michael Maier)
- [ASTERISK-29030] - res_rtp_asterisk: Additional RTP-frame (with wrong SSRC)
-                    gets inserted when switching from progress to established
-                    (Reported by Matthias Hensler)
- [ASTERISK-29407] - chan_local: Filtering audio formats should not occur on
-                    removed streams (Reported by Joshua C. Colp)
- [ASTERISK-29328] - translate.c: possible buffer overflow when upsampling
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29379] - Segfault - ast_channel_is_multistream (chan=0x0) at
-                    channel_internal_api.c:1590 (Reported by Ross Beer)
- [ASTERISK-29130] - prometheus: Crash when scraping bridge (Reported by
-                    Francisco Correia)
- [ASTERISK-29364] - res_rtp_asterisk: standard deviation miscalculation
-                    (Reported by Kevin Harwell)
- [ASTERISK-29373] - res_rtp_asterisk: Flash events are duplicated (Reported by
-                    N A)
- [ASTERISK-28356] - app_queue: CLI set ringinuse for realtime member not
-                    working (Reported by Michael)
- [ASTERISK-24434] - Fix differing usage of assignment operators in modules.conf
-                    (Reported by Rusty Newton)
- [ASTERISK-26614] - app_queue: updatecdr option in queues.conf does effectively
-                    nothing (Reported by Alexander Gonchiy)
- [ASTERISK-24631] - Incorrect description of option "context" in
-                    queues.conf.sample (Reported by Etienne Lessard)
- [ASTERISK-25358] - dateformat not read from logger.conf by remote console
-                    (Reported by Igor Liferenko)
- [ASTERISK-27542] - app_queue: When "queue show" CLI command is executed a
-                    crash occurs (Reported by Miguel Sanz)
- [ASTERISK-29215] - res_pjsip_session: NULL active_media_state topology caused
-                    asterisk crash (Reported by sungtae kim)
- [ASTERISK-29355] - app_queue: Queue member status message sent even if status
-                    doesn't change (Reported by Roman Pertsev)
- [ASTERISK-29035] - chan_local: Multistream support breaks T.38 faxing
-                    (Reported by Matthias Hensler)
- [ASTERISK-29354] - res_pjsip: Allow partial reloading of transports (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29348] - menuselect doesn't return errors in many cases (Reported by
-                    George Joseph)
- [ASTERISK-29352] - res_rtp_asterisk: Fix frame delivery time when SSRC changes
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29071] - app_confbridge: Memory rises when jitterbuffer enabled and
-                    muting over AMI occurs (Reported by Stefan Ruf)
- [ASTERISK-29329] - app_dial: DTMF to 'D' option gets duplicated if there are
-                    multiple progress events (Reported by N A)
- [ASTERISK-29306] - strings: Incorrect use of __attribute__((pure)) in
-                    ast_str_to_lower definition (Reported by Vitezslav Novy)
- [ASTERISK-29300] - res_rtp_asterisk: When native local bridging the remote
-                    SSRC becomes permanent (Reported by Sebastian Damm)
- [ASTERISK-29235] - res_pjsip_nat: Contact is rewritten on REGISTER responses
-                    with external_signaling_address (Reported by Brian Paboojian)
- [ASTERISK-29266] - ICE Role conflict with an unauthorized session (Reported by
-                    Salah Ahmed)
- [ASTERISK-29105] - chan_pjsip: 180 Ringing with SDP not changed into progress
-                    (Reported by Sebastian Damm)
- [ASTERISK-29297] - say: Y2021 problem – Asterisk cannot say year 2021 in
-                    Dutch (Reported by Jacek Konieczny)
- [ASTERISK-29315] - res_pjsip: re-registration gets stuck if setting initial
-                    auth credentials fails (Reported by Nick French)
- [ASTERISK-29312] - res_fax: asterisk fails to publish the Stasis and
-                    ReceiveFax status messages if the remote Station ID contains
-                    invalid UTF-8 characters (Reported by Alexei Gradinari)
- [ASTERISK-16799] - Callee declined when 'beep' audio file does not exist
-                    (Reported by IAMJames_)
- [ASTERISK-29313] - res_pjsip_refer: Segfault in progress notify (Reported by
-                    George Joseph)
- [ASTERISK-28452] - pjsip: of SDP is not incremented though SDP may be changed
-                    on reinvite without SDP offer (Reported by Michael Maier)
- [ASTERISK-29311] - res_odbc_transaction sets forcecommit default value based
-                    on isolation level instead of forcecommit (Reported by Jaco
-                    Kroon)
- [ASTERISK-29303] - pjsip: Re-invite occurs when it shouldn't (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-29293] - res_config_pgsql: Limit realtime_pgsql() to return one (no
-                    more) record (Reported by Boris P. Korzun)
- [ASTERISK-28369] - app_queue: Member device state "invalid" when second call
-                    is ringing and hint is used (Reported by Boolah)
- [ASTERISK-29287] - app.h: C++ compatibility broken (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29203] - res_pjsip_t38: Crash when changing state (Reported by
-                    Gregory Massel)
- [ASTERISK-29205] - res_rtp_asterisk: Asterisk crashes when making hold/unhold
-                    from webrtc client (Reported by Edvin Vidmar)
- [ASTERISK-29196] - res_pjsip: Segmentation fault (Reported by Mauri de Souza
-                    Meneguzzo (3CPlus))
- [ASTERISK-29280] - chan_sip: Allow peers without audio (text+video). (Reported
-                    by Alexander Traud)
- [ASTERISK-29265] - chan_sip: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29259] - channel: Allow text+video media streams, again. (Reported
-                    by Alexander Traud)
- [ASTERISK-29261] - res_pjsip: user=phone validation fail for isup numbers
-                    containing *# (Reported by Mark Petersen)
- [ASTERISK-29258] - chan_sip: Audio stream rejected, Other stream present:
-                    Invalid SDP. (Reported by Alexander Traud)
- [ASTERISK-29248] - res_pjsip_session: res sometimes uninitialized reported by
-                    compiler Clang. (Reported by Alexander Traud)
- [ASTERISK-29220] - After T38 reinvite response of 488 a subsequent G711
-                    reinvite is not processed correctly. Instead the previous T38
-                    session media is used (Reported by Robert Cripps)
- [ASTERISK-29229] - Stasis/messaging: text messages not dispatched to all
-                    subscribers when using generic subscription (Reported by Jean
-                    Aunis - Prescom)
- [ASTERISK-29238] - chan_sip: SDP: Offers without any enabled stream are
-                    accepted. (Reported by Alexander Traud)
- [ASTERISK-29237] - chan_sip: SDP: m=video is parsed even when disabled.
-                    (Reported by Alexander Traud)
- [ASTERISK-29222] - chan_sip: Hold/Resume an sRTP call on a video enabled
-                    user-agent. (Reported by Alexander Traud)
- [ASTERISK-29240] - chan_pjsip: Incoming PJSIP calls set global SIPDOMAIN
-                    instead of a channel variable (Reported by Ivan Poddubny)
- [ASTERISK-27902] - chan_pjsip isn't updating hangupcause on 4XX responses
-                    (Reported by George Joseph)
- [ASTERISK-28016] - PJSIP sends duplicate 183 Progress responses (Reported by
-                    Alex Hermann)
- [ASTERISK-28185] - chan_pjsip: Subsequent same responses are not stopped
-                    (Reported by Julien)
- [ASTERISK-29230] - pjsip: Asterisk goes crazy and massively spams logfile if
-                    registration can't be send (Reported by Michael Maier)
- [ASTERISK-29231] - pjsip: SIGSEGV in CLI if no trunk is registered (Reported
-                    by Michael Maier)
- [ASTERISK-29217] - LOCK() can grant the same lock to multiple channels
-                    spuriously (Reported by Jaco Kroon)
- [ASTERISK-28947] - Segmentation fault in mixmonitor_ds_destroy (Reported by
-                    Robert Sutton)
- [ASTERISK-29201] - Crash occurs when Transfer and execute Hangup before the
-                    Transfer result (Reported by Dan Cropp)
- [ASTERISK-29168] - Asterisk crashes during call transfer (Reported by Dalius
-                    Mockevicius)
- [ASTERISK-29210] - res_pjsip: Crash when examining transport (Reported by N GM
-                    )
- [ASTERISK-29191] - tel: URI in Diversion header causes crash (Reported by
-                    Mikhail Ivanov)
- [ASTERISK-28883] - Spyee information ist missing in ChanSpyStop AMI Event
-                    (Reported by Hendrik Wedhorn)
- [ASTERISK-29188] - null media causing the Asterisk crash (Reported by sungtae
-                    kim)
- [ASTERISK-29209] - Debug messages printed by scope trace might be missing
-                    newlines (Reported by Alexander Traud)
- [ASTERISK-29024] - pjsip: Route Header in Cancel request incorrectly set
-                    (Reported by Flole Systems)
- [ASTERISK-29211] - res_musiconhold: Segfault on realtime music on hold without
-                    entries (Reported by Nathan Bruning)
- [ASTERISK-29022] - Crash when manipulating PJSIP invite dlg ref counts
-                    (Reported by Sean Bright)
- [ASTERISK-29173] - Media cache URL requests allow infinite redirects (Reported
-                    by Sean Bright)
- [ASTERISK-29175] - res_pjsip_stir_shaken: Fix module description (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-29148] - AST_MODULE_INFO no, MODULEINFO depend (Reported by
-                    Alexander Traud)
- [ASTERISK-29165] - res_pjsip: malformed header Accept-Encoding in OPTIONS
-                    response (Reported by Alexander Greiner-Baer)
- [ASTERISK-28798] - [patch] chan_sip: TCP/TLS client without server. (Reported
-                    by Alexander Traud)
- [ASTERISK-29161] - Incorrect setup of recall channels (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29155] - app_queue: Deadlock between queues container and individual
-                    queues (Reported by George Joseph)
- [ASTERISK-28933] - res_pjsip.so fails to load when bundled pjproject is
-                    compiled without libssl (Reported by Walter Doekes)
- [ASTERISK-28825] - Any curl response checks out as valid even if 404 is
-                    returned. (Reported by dovid)
- [ASTERISK-29013] - res_pjsip: Asterisk doesn't stop sending invites (with
-                    auth) on 407 replies (Reported by Sebastian Damm)
- [ASTERISK-29142] - sip_to_pjsip.py: doesn't read globbed includes (Reported by
-                    Michael Newton)
- [ASTERISK-29144] - GCC Warnings with OPTIMIZE=-Og make (Reported by Alexander
-                    Traud)
- [ASTERISK-29146] - GCC Warnings: ‘%s’ directive argument is null.
-                    (Reported by Alexander Traud)
- [ASTERISK-29145] - GCC Warnings with OPTIMIZE=-Os make (Reported by Alexander
-                    Traud)
- [ASTERISK-29124] - res_pjsip: flow transport broken for outbound requests
-                    (Reported by Nick French)
- [ASTERISK-29136] - config: Sample features.conf incorrectly includes " around
-                    sound files (Reported by Benjamin M.)
- [ASTERISK-29123] - logger.conf.sample missing comment mark on line 115
-                    (Reported by Andrew Siplas)
- [ASTERISK-29109] - res_pjsip_session: Asterisk 18 does not progress calls due
-                    to codec negotiation after upgrading from Asterisk 16
-                    (Reported by Ross Beer)
- [ASTERISK-28430] - res_rtp_asterisk.c: FRACK!, Failed assertion errno != EBADF
-                    (Reported by under)
- [ASTERISK-29108] - resource_endpoints.c : Memory leak if endpoint not found
-                    (Reported by Jean Aunis - Prescom)
- [ASTERISK-29097] - res_pjsip_config_wizard: Crash when freeing string when
-                    failing to add extension (Reported by Vieri)
- [ASTERISK-26424] - app_voicemail: Undocumented behavior from VMSayName
-                    (Reported by Eric Smith)
- [ASTERISK-29051] - res_pjsip_sdp_rtp: Does not set correct values on RTP
-                    instance when "auto" DTMF is used (Reported by Sebastian
-                    Damm)
- [ASTERISK-29099] - res_musiconhold: Realtime MOH only loads a single entry
-                    (Reported by laszlovl)
- [ASTERISK-29091] - Crash when ast_translator_build_path fails (Reported by
-                    Jasper van der Neut)
- [ASTERISK-28311] - dsp: ast_dsp_silence_noise_with_energy wrong judgment of
-                    frame format (Reported by 周家建)
- [ASTERISK-29085] - func_curl: Segmentation fault when using CURL after setting
-                    httpheader CURLOPT (Reported by Péter Juhász)
- [ASTERISK-24329] - Music On Hold announcement cuts intro of music the first
-                    time it is played (Reported by Thomas Frederiksen)
- [ASTERISK-29089] - RTP Ports not cleared after hangup (Reported by Ross Beer)
- [ASTERISK-29081] - res_stasis: Add compare function for bridges moh container
-                    (Reported by Hajek Michal)
- [ASTERISK-28416] - Unable to get rtp codec payload code for slin (Reported by
-                    Brian J. Murrell)
- [ASTERISK-29014] - res_pjsip_session: Re-INVITE collisions aren't handled
-                    correctly (Reported by George Joseph)
- [ASTERISK-25665] - Duplicate logging in queue log for EXITEMPTY events
-                    (Reported by Ove Aursand)
- [ASTERISK-29043] - app_queue: Leave empty sometimes not recorded as abandoned
-                    (Reported by Kfir Itzhak)
- [ASTERISK-29042] - res_parking: Parker UUID is no longer copied (Reported by
-                    Misha Vodsedalek)
- [ASTERISK-28878] - chan_pjsip: PJSIP_MEDIA_OFFER Broken asterisk 16 (Reported
-                    by Joseph Ades)
- [ASTERISK-29046] - pbx: Deadlock when doing a reload, while simultaneously
-                    doing an ExtensionState on a pattern match hint that ends up
-                    adding an extension (Reported by Ramarajan)
- [ASTERISK-29040] - res_speech: Assertion on format (Reported by Nickolay V.
-                    Shmyrev)
- [ASTERISK-29001] - chan_pjsip does not process or forward 181 responses
-                    (Reported by Torrey Searle)
- [ASTERISK-27273] - app_voicemail: When a voicemail is marked as "Urgent", it
-                    is not sent by email/processed by the mailcmd command
-                    (Reported by Leandro Dardini)
- [ASTERISK-29034] - Lastpause of realtime members is reseting (Reported by
-                    Evandro César Arruda)
- [ASTERISK-29033] - res_pjsip_session: Aggressively terminates session on
-                    failed re-INVITE (Reported by Joshua C. Colp)
- [ASTERISK-28974] - res_rtp_asterisk: T.140 messages have appended RTP string
-                    to each message block. (Reported by Thomas Johnson)
- [ASTERISK-29011] - chan_sip: ToHost property not cleared on reload (Reported
-                    by Dennis)
- [ASTERISK-29021] - [patch] Fix VERSION(ASTERISK_VERSION_NUM) on certified
-                    versions (Reported by cmaj)
- [ASTERISK-28927] - Asterisk crash in music on hold (Reported by David
-                    Cunningham)
- [ASTERISK-28973] - Malformed IP address in SDP of 2nd SIP timer triggered
-                    INVITE when NAT is active (UDP transport with
-                    external_media_address) (Reported by Michael Neuhauser)
- [ASTERISK-28995] - res_pjsip_registrar: Expires on statically configured
-                    contacts is not correct (Reported by tootai)
- [ASTERISK-28987] - BridgeCreated ARI event shows wrong video_mode info
-                    (Reported by sungtae kim)
- [ASTERISK-28978] - acl: named_acl rule misconfiguration results in segfault on
-                    reading rule from realtime (Reported by Andrew Yager)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
- [ASTERISK-29501] - ARI - Stasis Playback doesn't hangup call when processing a
-                    list of invalid files (Reported by Andre Barbosa)
- [ASTERISK-29464] - ARI - PlaybackFinish skip error events (Reported by Andre
-                    Barbosa)
- [ASTERISK-29450] - Allow setting channel variables using Originate application
-                    (Reported by N A)
- [ASTERISK-29460] - Recognize application/hook-flash in PJSIP (Reported by N A)
- [ASTERISK-29459] - Missing configuration from PJSIP to SIP conversion script
-                    (Reported by N A)
- [ASTERISK-29434] - Asterisk reveals pjproject version in STUN packets
-                    (Reported by Jeremy Lainé)
- [ASTERISK-29380] - Add Flash AMI event to handle flash events (Reported by N
-                    A)
- [ASTERISK-29349] - Silent voicemail option is not completely silent (Reported
-                    by N A)
- [ASTERISK-29339] - loader: Let's output warnings for deprecated modules!
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29337] - menuselect: Add ability to set deprecated in and removed in
-                    versions for modules (Reported by Joshua C. Colp)
- [ASTERISK-29335] - xml: Embed module information into core XML documentation.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29336] - documentation: Fix inconsistent support levels (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29321] - sorcery: Add support for more intelligent reloading.
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29325] - res_pjsip_registrar: Include source IP address and port in
-                    log messages (Reported by Joshua C. Colp)
- [ASTERISK-29326] - asterisk: Update copyright/company (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29244] - Add MixMonitorStart / Stop / Mute AMI events (Reported by
-                    Sébastien Duthil)
- [ASTERISK-29252] - TRANSFERSTATUSPROTOCOL variable to report Transfer (REFER)
-                    failure SIP code (Reported by Dan Cropp)
- [ASTERISK-29262] - Support of various URL-schemes by MoH (Reported by Boris P.
-                    Korzun)
- [ASTERISK-28549] - Two repeated 183 (Reported by Gant Liu)
- [ASTERISK-29216] - contrib: systemd asterisk service for centos8 or other
-                    newer linux versions (Reported by Mark Petersen)
- [ASTERISK-29143] - res_http_media_cache: HTTP media cache stored hardcoded in
-                    /tmp (Reported by laszlovl)
- [ASTERISK-29118] - VoiceMail() should have an option to play greetings as
-                    Early Media (Reported by Juan Carlos Castro y Castro)
- [ASTERISK-29054] - Logger: Add debug logging categories (Reported by Kevin
-                    Harwell)
- [ASTERISK-29083] - Do not build chan_sip by default as it is now deprecated
-                    (Reported by Sean Bright)
- [ASTERISK-29056] - Increase reg_server column size for ps_contacts table
-                    realtime (Reported by sungtae kim)
- [ASTERISK-29055] - Create a Bridge with video_single mode (Reported by sungtae
-                    kim)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="cd35b546081c9c2fe42ecde13e46b2e7ea831e10bb9f62c3ce249cd30ceb26b9" name="asterisk" arch="src">
  <version epoch="0" ver="19.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.1.0-1" date="1639051200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors (Reported
-                    by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined
-                    (Reported by Alexander Traud)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="b5567c93859d1f58dfdf2a5e64c0a9864a012609a4ee723981eebfff43ec6369" name="asterisk" arch="src">
  <version epoch="0" ver="19.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.0-1" date="1644494400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="6b12792c324dc43aa54c28b898f474e6c7f7e6c89837a83faffbbd2a1a5c5c95" name="asterisk" arch="src">
  <version epoch="0" ver="19.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.2.1-1" date="1646395200">- Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.24.1, 18.10.1, 19.2.1 and 16.8-cert13.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- - AST-2022-004: pjproject: integer underflow on STUN message
-   The header length on incoming STUN messages that contain an ERROR-CODE
-   attribute is not properly checked. This can result in an integer underflow.
-   Note, this requires ICE or WebRTC support to be in use with a malicious
-   remote party.
-
- - AST-2022-005: pjproject: undefined behavior after freeing a dialog set
-   When acting as a UAC, and when placing an outgoing call to a target that
-   then forks Asterisk may experience undefined behavior (crashes, hangs,
-   etc…) after a dialog set is prematurely freed.
-
- - AST-2022-006: pjproject: unconstrained malformed multipart SIP message
-   If an incoming SIP message contains a malformed multi-part body an out of
-   bounds read access may occur, which can result in undefined behavior. Note,
-   it’s currently uncertain if there is any externally exploitable vector
-   within Asterisk for this issue, but providing this as a security issue out
-   of caution.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.24.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.10.1
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.2.1
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert13
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-004.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-005.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-006.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="1947814f34dc87cb073d04527394c238f0b0b68faed287559ff092d59b20b182" name="asterisk" arch="src">
  <version epoch="0" ver="19.6.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.0-1" date="1648123200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29945] - pjproject: Security fixes for things (Reported by Kevin
-                    Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
</package>
<package pkgid="d766686b1d1b2ac40d266633e9bb505c20423227e891d3a9a0061b73547d08ed" name="asterisk" arch="src">
  <version epoch="0" ver="19.6.0" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-1" date="1648555200">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
</package>
<package pkgid="054970ae16e646f9db11e6607df6cb6400a54bbc29ae3fdb9421b40520151b81" name="asterisk" arch="src">
  <version epoch="0" ver="19.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.1-2" date="1649073600">- Added mediasec support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="99ee7a56bc754629d28409ba56d5c83e91a8be3da8fdaa71f8e67bb52093db7c" name="asterisk" arch="src">
  <version epoch="0" ver="19.7.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="30da043637396664cddcb7010160091e9bf175a6929c58744bb613ddf9ae0b84" name="asterisk" arch="src">
  <version epoch="0" ver="19.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="f8a0ed094d0a609a65fdcc5d9f2e6419df11cccea7c5e5d58791772bd6c55031" name="asterisk" arch="src">
  <version epoch="0" ver="19.8.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.8.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.8.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by
-                    N A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.8.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.8.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 19.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/19.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 19.8.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/19.8.0...19.8.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-19.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- bundled_pjproject: Backport 2 SSL patches from upstream
- bundled_pjproject: Backport security fixes from pjproject 2.13.1
- apply_patches: Sort patch list before applying
-
- User Notes:
- -----------
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #188: [improvement]:  pjsip: Upgrade bundled version to pjproject 2.13.1 #187
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying
- #194: [bug]: Segfault/double-free in bundled pjproject using TLS transport</changelog>
</package>
<package pkgid="c7147ef1ce26611a81bbe1d4a6e7e9d0f702c743f4d13aad06d98c10ee646773" name="asterisk" arch="src">
  <version epoch="0" ver="20.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.2-1" date="1649937600">- The Asterisk Development Team would like to announce security releases for
- Asterisk 16, 18 and 19, and Certified Asterisk 16.8. The available releases
- are released as versions 16.25.2, 18.11.2, 19.3.2 and 16.8-cert14.
-
- These releases are available for immediate download at
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases
-
- The following security vulnerabilities were resolved in these versions:
-
- AST-2022-001: res_stir_shaken: resource exhaustion with large files When using
- STIR/SHAKEN, it’s possible to download files that are not certificates.
- These files could be much larger than what you would expect to download.
-
- AST-2022-002: res_stir_shaken: SSRF vulnerability with Identity header When
- using STIR/SHAKEN, it’s possible to send arbitrary requests like GET to
- interfaces such as localhost using the Identity header.
-
- AST-2022-003: func_odbc: Possible SQL Injection Some databases can use
- backslashes to escape certain characters, such as backticks. If input is
- provided to func_odbc which includes backslashes it is possible for func_odbc
- to construct a broken SQL query and the SQL query to fail.
-
- For a full list of changes in the current releases, please see the ChangeLogs:
-
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-16.25.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.11.2
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-19.3.2
- https://downloads.asterisk.org/pub/telephony/certified-asterisk/releases/ChangeLog-certified-16.8-cert14
-
- The security advisories are available at:
-
- https://downloads.asterisk.org/pub/security/AST-2022-001.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-002.pdf
- https://downloads.asterisk.org/pub/security/AST-2022-003.pdf
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="92a3dae244eff15414bab8473b472661f0399fccc4146d1452261d29e91aea8b" name="asterisk" arch="src">
  <version epoch="0" ver="20.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.3.3-1" date="1650974400">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.3.3.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.3.3 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.3.3
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="df112f087cbfe33279840e16c0e49925dd5dc1b7ff84949307b1674e7f159625" name="asterisk" arch="src">
  <version epoch="0" ver="20.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.0-1" date="1652356800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="e6a2ebf66a7b4ad1eef353edf5b6df76d76dda677261433232a4489664d1bd36" name="asterisk" arch="src">
  <version epoch="0" ver="20.10.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="5edb9c6b3b3097b5eb9d6e782ebbbb7e4c68c23ae2699f17f4ce36bda5cfe093" name="asterisk" arch="src">
  <version epoch="0" ver="20.11.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="268326e21c8b80a1cce5318fa5620ad058d09240c7c53d2612032d39cbff2540" name="asterisk" arch="src">
  <version epoch="0" ver="20.11.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="9c2627775ade15d39fb7af070363c8526321c32d04c2ec94f84d41b739af0b2a" name="asterisk" arch="src">
  <version epoch="0" ver="20.12.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="863e087b0dbbc5f089d1f750a6fe3501d162476aa2af944744f76ca6c1918deb" name="asterisk" arch="src">
  <version epoch="0" ver="20.13.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="320a613f273bdbed1ea4876b273157d2f76be6f3a988994524684df0958ebeae" name="asterisk" arch="src">
  <version epoch="0" ver="20.14.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="3943d327ad5cef11b48028879dc56201cecc09663f1f7982e942d7acd715ac71" name="asterisk" arch="src">
  <version epoch="0" ver="20.14.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------

- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-  - [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-  - [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Upgrade Notes:
- --------------
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ------------------------
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-  - res_pjsip_messaging.c: Mask control characters in received From display name
-  - asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author:   George Joseph
-             Date:     2025-05-19
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-                       has been added that, when set, will prevent remote
-                       consoles from executing shell commands using the '!'
-                       prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author:   George Joseph
-                        Date:     2025-03-24
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
</package>
<package pkgid="ab48a9fe95a683c5052dd940fd255817d5af500e6f63fb820a393e897811dad9" name="asterisk" arch="src">
  <version epoch="0" ver="20.15.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.10.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.10.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.10.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.10.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.10.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.10.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.3...20.10.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.10.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 9
- Issues Resolved: 18
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (10)
- Jaco Kroon: (1)
- Jean-Denis Girard: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.11.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.11.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 40
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like
-               `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.11.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 20.11.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.11.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.11.1
-
- Change Log for Release asterisk-20.11.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.0...20.11.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
-  manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-  Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.12.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.12.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.12.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.12.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.12.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.12.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.11.1...20.12.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.12.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.13.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.13.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.13.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.13.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.13.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.13.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.12.0...20.13.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.13.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (11)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.14.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.14.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.13.0...20.14.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.14.1-1" date="1747915200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.14.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.14.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.14.1
-
- Change Log for Release asterisk-20.14.1
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.14.1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.0...20.14.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.14.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------

- Commits: 2
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 2
-  - [GHSA-2grh-7mhv-fcfw](https://github.com/asterisk/asterisk/security/advisories/GHSA-2grh-7mhv-fcfw):
-    Using malformed From header can forge identity with ";" or NULL in name
-    portion
-  - [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- Upgrade Notes:
- --------------
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ------------------------
-
- Closed Issues:
- --------------
-
- !GHSA-2grh-7mhv-fcfw: Using malformed From header can forge identity with ";"
-                       or NULL in name portion
- !GHSA-c7p6-7mvq-8jq2: cli_permissions.conf: deny option does not work for
-                       disallowing shell commands
-
- Commits By Author:
- ------------------
-
- George Joseph (2):
-  - res_pjsip_messaging.c: Mask control characters in received From display name
-  - asterisk.c: Add option to restrict shell access from remote consoles.
-
- Commit List:
- ------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
- res_pjsip_messaging.c: Mask control characters in received From display name
-
- Commit Details:
- ---------------
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             Author:   George Joseph
-             Date:     2025-05-19
-             UserNote: A new asterisk.conf option 'disable_remote_console_shell'
-                       has been added that, when set, will prevent remote
-                       consoles from executing shell commands using the '!'
-                       prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- res_pjsip_messaging.c: Mask control characters in received From display name
-                        Author:   George Joseph
-                        Date:     2025-03-24
-                        Incoming SIP MESSAGEs will now have their From header's
-                        display name sanitized by replacing any characters &lt; 32
-                        (space) with a space.
-                        Resolves: #GHSA-2grh-7mhv-fcfw</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.15.0-1" date="1752753600">- The Asterisk Development Team would like to announce the release of
- asterisk-20.15.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.15.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.15.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.15.0
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.15.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.14.1...20.15.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.15.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 29
- Commit Authors: 14
- Issues Resolved: 19
- Security Advisories Resolved: 1
-  - [GHSA-c7p6-7mvq-8jq2](https://github.com/asterisk/asterisk/security/advisories/GHSA-c7p6-7mvq-8jq2):
-    cli_permissions.conf: deny option does not work for disallowing shell
-    commands
-
- User Notes:
- -----------
-
- res_stir_shaken.so: Handle X5U certificate chains.
-                     The STIR/SHAKEN verification process will now load a full
-                     certificate chain retrieved via the X5U URL instead of
-                     loading only the end user cert.
-
- res_stir_shaken: Add "ignore_sip_date_header" config option.
-                  A new STIR/SHAKEN verification option "ignore_sip_date_header"
-                  has been added that when set to true, will cause the
-                  verification process to not consider a missing or invalid SIP
-                  "Date" header to be a failure. This will make the IAT the sole
-                  "truth" for Date in the verification process. The option can
-                  be set in the "verification" and "profile" sections of
-                  stir_shaken.conf. Also fixed a bug in the port match logic.
-                  Resolves: #1251
-                  Resolves: #1271
-
- app_record: Add RECORDING_INFO function.
-             The RECORDING_INFO function can now be used to retrieve the
-             duration of a recording.
-
- app_queue: queue rules – Add support for QUEUE_RAISE_PENALTY=rN to raise penal..
-            This change introduces QUEUE_RAISE_PENALTY=rN, allowing selective
-            penalty raises only for members whose current penalty is within the
-            [min_penalty, max_penalty] range. Members with lower or higher
-            penalties are unaffected. This behavior is backward-compatible with
-            existing queue rule configurations.
-
- res_odbc: cache_size option to limit the cached connections.
-           New cache_size option for res_odbc to on a per class basis limit the
-           number of cached connections. Please reference the sample
-           configuration for details.
-
- res_odbc: cache_type option for res_odbc.
-           When using res_odbc it should be noted that back-end connections to
-           the underlying database can now be configured to re-use the cached
-           connections in a round-robin manner rather than repeatedly re-using
-           the same connection. This helps to keep connections alive, and to
-           purge dead connections from the system, thus more dynamically
-           adjusting to actual load. The downside is that one could keep too
-           many connections active for a longer time resulting in resource also
-           begin consumed on the database side.
-
- ARI Outbound Websockets
-      Asterisk can now establish websocket sessions _to_ your ARI applications
-      as well as accepting websocket sessions _from_ them. Full details:
-      http://s.asterisk.net/ari-outbound-ws
-
- res_websocket_client: Create common utilities for websocket clients.
-                       A new module "res_websocket_client" and config file
-                       "websocket_client.conf" have been added to support
-                       several upcoming new capabilities that need common
-                       websocket client configuration.
-
- asterisk.c: Add option to restrict shell access from remote consoles.
-             A new asterisk.conf option 'disable_remote_console_shell' has been
-             added that, when set, will prevent remote consoles from executing
-             shell commands using the '!' prefix.
-             Resolves: #GHSA-c7p6-7mvq-8jq2
-
- sig_analog: Add Call Waiting Deluxe support.
-             Call Waiting Deluxe can now be enabled for FXS channels by enabling
-             its corresponding option.
-
- Upgrade Notes:
- --------------
-
- jansson: Upgrade version to jansson 2.14.1
-          jansson has been upgraded to 2.14.1. For more information visit
-          jansson Github page:
-          https://github.com/akheron/jansson/releases/tag/v2.14.1
-          Resolves: #1178
-
- Alternate Channel Storage Backends
-      With this release, you can now select an alternate channel storage backend
-      based on C++ Maps. Using the new backend may increase performance and
-      reduce the chances of deadlocks on heavily loaded systems. For more
-      information, see http://s.asterisk.net/dc679ec3
-
- Commit Authors:
- ---------------
-
- George Joseph: (10)
- Itzanh: (1)
- Jaco Kroon: (2)
- Joe Searle: (1)
- Michal Hajek: (1)
- Mike Bradeen: (2)
- Mkmer: (1)
- Nathan Monfils: (1)
- Naveen Albert: (3)
- Phoneben: (1)
- Sean Bright: (2)
- Stanislav Abramenkov: (1)
- Sven Kube: (2)
- Thomas B. Clark: (1)</changelog>
</package>
<package pkgid="e0ed96b74fee6d94afe00e4b049089003fe22a692116203fdffa19fd684ee393" name="asterisk" arch="src">
  <version epoch="0" ver="20.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.4.1-1" date="1652961600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.4.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.4.1 resolves an issue reported by the community and
- would have not been possible without your participation.
-
- Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.4.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="4806e6ec33aee6a271f7b5692047ce3e965a06973788e0e5079a361deeaf05f9" name="asterisk" arch="src">
  <version epoch="0" ver="20.2.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.5.0-1" date="1655985600">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.5.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.5.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch]Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.5.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
</package>
<package pkgid="5ab4dbdf8c2e7714ff628500928fbdfad165b00f63a5246ef5e00fa761873317" name="asterisk" arch="src">
  <version epoch="0" ver="20.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-1" date="1660824000">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.6.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.6.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.6.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824001">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
</package>
<package pkgid="d00c01c03914f2e877bfac4135c6cb1fe0ad164377526c4ae51353e20cf4c508" name="asterisk" arch="src">
  <version epoch="0" ver="20.3.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-2" date="1660824000">- Added Cisco CallManager support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
</package>
<package pkgid="a62bfb7b43a59880102d4ad7c070a98516d52144f127ff6004904987652a8414" name="asterisk" arch="src">
  <version epoch="0" ver="20.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.6.0-3" date="1666008000">- Added geolocation subpackage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
</package>
<package pkgid="222b148648b4814bcdd0c934d9008f355dd3e158cf38c9c945966a01574deea4" name="asterisk" arch="src">
  <version epoch="0" ver="20.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 19.7.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 19.7.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 19.7.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180801">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
</package>
<package pkgid="4f2e75fc5967db7398c8dc357826802dcd9eade357cadc33e55f30c1d9c07ca8" name="asterisk" arch="src">
  <version epoch="0" ver="20.5.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="94f864be34a4a9eaa35d04fa8684765cc8944802ca1443c5f26afee4468b2e43" name="asterisk" arch="src">
  <version epoch="0" ver="20.5.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="096e6fd5166f2f13ff65676e3002349f9c1f3e43d015312cad299d0b52a6eb23" name="asterisk" arch="src">
  <version epoch="0" ver="20.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="c33c412605aa58596ccc62e74ebf206c340051ff5d7f58b42fb676ca686f8322" name="asterisk" arch="src">
  <version epoch="0" ver="20.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="2f758cc2e81ee1b4194a22bed125fe2bbe87524f833c49cb57107c29732eb07b" name="asterisk" arch="src">
  <version epoch="0" ver="20.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="8fffbc586c99efc16d43de0225d6af7122d87a96644b02621467b40c3eb858c1" name="asterisk" arch="src">
  <version epoch="0" ver="20.8.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="8cb3b8fcc5a2095e92fcacea77c55160029ef48a7580bd9718fefdbb7164e540" name="asterisk" arch="src">
  <version epoch="0" ver="20.9.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="4344aaa009438a633ac77bbf0cb76dd204ff884c76d0471a0d604585e1bc1880" name="asterisk" arch="src">
  <version epoch="0" ver="20.9.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="52ac35dca1bc9817cff9e3d904e41bad7243c2f37e17865269363569371b964f" name="asterisk" arch="src">
  <version epoch="0" ver="20.9.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="5fbe8bc00e528163e2c78cf3c741020608e81eaedf251c13e76be1ed96f3dec7" name="asterisk" arch="src">
  <version epoch="0" ver="20.9.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.5.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- [Path traversal via AMI GetConfig allows access to outside files]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
- initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- [PJSIP logging allows attacker to inject fake Asterisk log entries ]
- (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
- 'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-20.5.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.0...20.5.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_header_funcs: Duplicate new header value, don't copy.
- res_pjsip: disable raw bad packet logging
- res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.2-1" date="1702555201">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.1...20.5.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.6.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.5.2...20.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- Update config.yml
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-              built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.7.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-20.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.6.0...20.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- alembic: Synchronize alembic heads between supported branches.
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail: add NoOp alembic script to maintain sync
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------
-
- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-       available on parsing failure
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on spied
-                      channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.7.0...20.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 44
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport
-                                        address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (8)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.8.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.8.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.8.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.8.1
-
- Change Log for Release asterisk-20.8.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.8.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.0...20.8.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.8.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.0
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.0
-
- This release resolves issues reported by the community and would have not been
-  possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.8.1...20.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message
-                          and greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes') to control
-                 whether the Restricted Caller ID will be stored in the queue
-                 log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-20.9.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.9.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.0...20.9.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   - voicemail.conf.sample: Fix ':' comment typo
-   - rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
-  rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-  voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls
-                    ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.9.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.2
-
- Change Log for Release asterisk-20.9.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.1...20.9.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-   Write=originate, is sufficient permissions for code execution / System()
-   dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   - manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io- 20.9.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 20.9.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.9.3
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 20.9.3
-
- Change Log for Release asterisk-20.9.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.9.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.9.2...20.9.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.9.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-    A malformed Contact or Record-Route URI in an incoming SIP request can cause
-    Asterisk to crash when res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
-
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="3cd705dad0d34f0f196b0d5f49726fff3d1d30ec28c1a52f2c6d7f23faec0ad1" name="asterisk" arch="src">
  <version epoch="0" ver="21.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.0-1" date="1666180800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.0.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.0.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Deprecations made in this release:
- ----------------------------------
-
- [ASTERISK-29601] - moduleinfo: Add replacement module information (Reported by
-                    N A)
- [ASTERISK-29602] - res_monitor: Disable building by default. (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29600] - muted: Remove deprecated application (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29599] - conf2ael: Remove deprecated application (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29598] - res_config_sqlite: Remove deprecated module (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29597] - chan_vpb: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29596] - chan_misdn: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29595] - chan_nbs: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29594] - chan_phone: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29593] - chan_oss: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29592] - cdr_syslog: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29591] - app_dahdiras: Remove deprecated module (Reported by Joshua
-                    C. Colp)
- [ASTERISK-29590] - app_nbscat: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29589] - app_image: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29588] - app_url: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29587] - app_fax: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29586] - app_ices: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29585] - app_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29584] - cdr_mysql: Remove deprecated module (Reported by Joshua C.
-                    Colp)
- [ASTERISK-29548] - app_meetme: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29549] - app_osploop: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29550] - chan_alsa: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29551] - chan_mgcp: Deprecated in 19, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29552] - chan_skinny: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29553] - res_pktccops: Deprecated in 19, to be removed in 21
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29558] - app_macro: Deprecated in 16, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29567] - chan_sip: Deprecated in 17, to be removed in 21 (Reported
-                    by Joshua C. Colp)
- [ASTERISK-29572] - res_monitor: Deprecated in 16, to be removed in 21
-                    (Reported by Joshua C. Colp)
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-29476] - res_stir_shaken: Blind SSRF vulnerabilities (Reported by
-                    Clint Ruoho)
- [ASTERISK-29872] - res_stir_shaken: Resource exhaustion with large files
-                    (Reported by Benjamin Keith Ford)
- [ASTERISK-29838] - ${SQL_ESC()} not correctly escaping a terminating \
-                    (Reported by Leandro Dardini)
- [ASTERISK-29415] - Crash in PJSIP TLS transport (Reported by Andrew Yager)
- [ASTERISK-29381] - chan_pjsip: Remote denial of service by an authenticated
-                    user (Reported by Ivan Poddubny)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-30037] - Add test support to calling external processes (Reported by
-                    Philip Prindeville)
- [ASTERISK-30161] - locks: add AMI event for deadlock (Reported by N A)
- [ASTERISK-30211] - app_confbridge: Add end_marked_any option (Reported by N A)
- [ASTERISK-30186] - res_pjsip: Add support for reloading TLS certificate and
-                    key information (Reported by Joshua C. Colp)
- [ASTERISK-29899] - features: Add advanced transfer initiation options
-                    (Reported by N A)
- [ASTERISK-30136] - db: Add AMI action to retrieve all keys beginning with a
-                    prefix (Reported by N A)
- [ASTERISK-30000] - chan_dahdi: Add POLARITY function (Reported by N A)
- [ASTERISK-30062] - cli: Add CLI command to execute a dialplan app (Reported by
-                    N A)
- [ASTERISK-29999] - pjsip: Get information from 200 OK INVITE reply headers
-                    (Reported by José Lopes)
- [ASTERISK-30061] - pbx: Add pbx helper application (Reported by N A)
- [ASTERISK-30063] - app_voicemail: Add option to prevent deletion of messages
-                    (Reported by N A)
- [ASTERISK-30087] - res_parking: Add music on hold override option (Reported by
-                    N A)
- [ASTERISK-29965] - res_pjsip_outbound_registration: Make max registration
-                    delay configurable (Reported by N A)
- [ASTERISK-30036] - app_confbridge: Add CONFBRIDGE_CHANNELS function (Reported
-                    by N A)
- [ASTERISK-29931] - Option to allow a user to not hear the join sound on enter
-                    but everyone else can (Reported by Michael Cargile)
- [ASTERISK-29968] - func_db: Add a function to return cardinality of keys at
-                    prefix (Reported by N A)
- [ASTERISK-29486] - Hint-like extension value lookup function without device
-                    state (Reported by N A)
- [ASTERISK-29941] - chan_pjsip: Add ability to send flash events (Reported by N
-                    A)
- [ASTERISK-29820] - cli: Add command to evaluate a function (Reported by N A)
- [ASTERISK-29876] - app_queue: Add music on hold option (Reported by N A)
- [ASTERISK-29840] - func_channel: Add LASTCONTEXT and LASTEXTEN fields
-                    (Reported by N A)
- [ASTERISK-29853] - ami: Allow events to be globally disabled (Reported by N A)
- [ASTERISK-29808] - cdr: allow disabling CDR by default (Reported by N A)
- [ASTERISK-29830] - ami: Add AMI event for Wink (Reported by N A)
- [ASTERISK-29802] - app_sf: Add full tech-agnostic SF support (Reported by N A)
- [ASTERISK-29759] - app_sendtext: Add ReceiveText application (Reported by N A)
- [ASTERISK-29706] - func_json: Add JSON parsing function (Reported by N A)
- [ASTERISK-29720] - res_tonedetect: Add call progress tone detection (Reported
-                    by N A)
- [ASTERISK-18069] - [patch] app_queue Add Login Time and Last Paused Times to
-                    Queue Members (Reported by Jamuel Starkey)
- [ASTERISK-29656] - Add CHANNEL_EXISTS function (Reported by N A)
- [ASTERISK-29496] - Add SendMF application (Reported by N A)
- [ASTERISK-29627] - Add STRBETWEEN function (Reported by N A)
- [ASTERISK-29628] - Add file and directory functions (Reported by N A)
- [ASTERISK-29531] - Add SAYFILES function (Reported by N A)
- [ASTERISK-29546] - Add tone detection module (Reported by N A)
- [ASTERISK-18454] - Option for Read to be able to accept # (Reported by Sta
-                    Retji)
- [ASTERISK-29542] - Add audio scrambler (Reported by N A)
- [ASTERISK-29478] - Function to drop frames in the TX or RX directions
-                    (Reported by N A)
- [ASTERISK-29389] - Add PJSIP_HEADERS() and ability to read header by pattern
-                    (Reported by Igor Goncharovsky)
- [ASTERISK-29477] - Function to asynchronously store digits dialed (Reported by
-                    N A)
- [ASTERISK-11] - AGI channel_status failure (Reported by bbawkon)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: ...may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30215] - Inbound SIP INVITE with Geo Location causing a Segmentation
-                    Fault (Reported by Dan Cropp)
- [ASTERISK-30135] - [res_musiconhold] Allows the moh only for the answered call
-                    (Reported by sungtae kim)
- [ASTERISK-26894] - pjsip should support tel uri scheme (Reported by Gergely
-                    Dömsödi)
- [ASTERISK-30210] - func_frame_trace: Channel masquerade triggers assertion
-                    (Reported by N A)
- [ASTERISK-30190] - res_geolocation: GEOLOC_PROFILE isn't returning correct
-                    values on incoming channel (Reported by George Joseph)
- [ASTERISK-29185] - chan_pjsip: Endpoint: allow = all is broken. (Reported by
-                    Alexander Traud)
- [ASTERISK-30192] - res_tonedetect: fix typo for frametype (Reported by N A)
- [ASTERISK-29453] - alembic: incoming_call_offer_pref and
-                    outgoing_call_offer_pref missing in "ps_endpoints" table
-                    (Reported by Daniel Thümen)
- [ASTERISK-26826] - testsuite: Add support for Python 3 (Reported by Joshua C.
-                    Colp)
- [ASTERISK-30167] - res_geolocation: Refactor for issues found by users
-                    (Reported by George Joseph)
- [ASTERISK-28422] - Memory Leak in Confbridge menu (Reported by Ted G)
- [ASTERISK-29917] - ami: FilterList action doesn't exist (Reported by N A)
- [ASTERISK-30020] - ConfbridgeListRooms Event Not Documented (Reported by
-                    Michael Cargile)
- [ASTERISK-30018] - app_meetme: MeetmeList AMI event not documented (Reported
-                    by Michael Cargile)
- [ASTERISK-30151] - Documentation doesn't include info about "field", a 3rd
-                    required parameter. (Reported by Chris Young)
- [ASTERISK-29966] - pbx_variables: ast_str_strlen can be wrong (Reported by N
-                    A)
- [ASTERISK-29905] - OSX: bininstall launchd issue on cross-platfrom build
-                    (Reported by Sergey V. Lobanov)
- [ASTERISK-30137] - manager: Global disabled event filtered is incomplete
-                    (Reported by N A)
- [ASTERISK-30109] - res_pjsip: no contact-status AMI event on register of
-                    prune-on-boot contact that uses the same URI as before
-                    Asterisk restart (Reported by Michael Neuhauser)
- [ASTERISK-29991] - chan_dahdi, callerid: Caller ID does not honor presentation
-                    (Reported by N A)
- [ASTERISK-30126] - Spelling mistake in configs/samples/queues.conf.sample
-                    (Reported by Sam Banks)
- [ASTERISK-30029] - build: Git security vulnerability fix is sad with our
-                    accessing git as root during "make install" (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29907] - res_pjsip, app_confbridge: Video call through ConfBridge
-                    with normal endpoints causes infinite loop/crash (Reported by
-                    N A)
- [ASTERISK-30138] - Compile failure in res_geolocation/geoloc_eprofile.c when
-                    optimization is enabled (Reported by George Joseph)
- [ASTERISK-30096] - cel_odbc: Column type 9 (field 'cdr:cel:eventtime') is
-                    unsupported at this time (Reported by Morvai Szabolcs)
- [ASTERISK-30083] - chan_iax2: Optional dependency on openssl/res_crypto is now
-                    mandatory (Reported by Dmitry Melekhov)
- [ASTERISK-30099] - test_aeap_transport: transport_connect_fail sporadically
-                    causes failure (Reported by Kevin Harwell)
- [ASTERISK-30123] - features: Update automixmon documentation to reflect
-                    reality (Reported by Trevor Peirce)
- [ASTERISK-30117] - pbx_lua: Remove compiler warnings (Reported by Boris P.
-                    Korzun)
- [ASTERISK-30101] - res_prometheus: Optional load
-                    res_pjsip_outbound_registration.so (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29989] - app_dial, chan_dahdi: DIALSTATUS is inconsistent for busy
-                    (Reported by N A)
- [ASTERISK-30001] - db: Removing nonexistent entries shows "Database entry
-                    removed" (Reported by N A)
- [ASTERISK-29822] - cli: Typing \? freezes the CLI permanently with remote
-                    console (Reported by N A)
- [ASTERISK-30115] - app_dial: Allow hook flashes to propogate on outbound dials
-                    (Reported by N A)
- [ASTERISK-30106] - res_calendar_icalendar: Microsoft online ICS calendars no
-                    longer work (Reported by N A)
- [ASTERISK-30075] - say: Abort if channel hangs up during playback (Reported by
-                    N A)
- [ASTERISK-30072] - res_pjsip: allow TLS verification of wildcard cert-bearing
-                    servers (Reported by Kevin Harwell)
- [ASTERISK-30097] - console: Recent documentation changes for connecting to
-                    remote console are inconsistent (Reported by Matthias
-                    Hensler)
- [ASTERISK-30043] - Wrong party is disconnected when hook-flashing on 3-way
-                    bridge (Reported by Josh Alberts)
- [ASTERISK-29603] - res_pjsip: UPDATE/re-INVITE not sent when "timers=always"
-                    is specified in pjsip.conf (Reported by Ray Crumrine)
- [ASTERISK-30092] - DateTime application: wrong inflection for one o'clock in
-                    German (Reported by Christof Efkemann)
- [ASTERISK-30064] - pbx: iax2 switch causes crash due to deadlock and assertion
-                    (Reported by N A)
- [ASTERISK-30039] - cli: Targeted debug on startup deadlocks and creates
-                    unstable system (Reported by N A)
- [ASTERISK-29981] - res_calendar: Asterisk crashes when starting, and will not
-                    run (Reported by N A)
- [ASTERISK-30051] - res_pjsip: No video after un-hold with moh_passthrough=yes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-24601] - [patch] Missing RFC4235 tags and attributes in PJSIP NOTIFY
-                    event: dialog XML body (Reported by Marco Paland)
- [ASTERISK-30059] - menuselect: libxml include fails under Gentoo (Reported by
-                    waltermoeller)
- [ASTERISK-30060] - loader: format warnings in dev mode (Reported by N A)
- [ASTERISK-30065] - pjsip: Open Websocket connection is not reused for outgoing
-                    requests (Reported by LA)
- [ASTERISK-30042] - res_pjsip_transport_websocket: Registration over websocket
-                    returns a rewritten contact (Reported by Thomas Guebels)
- [ASTERISK-29993] - chan_dahdi: Operator control option borks both lines
-                    involved on callee disconnect (Reported by N A)
- [ASTERISK-30044] - GCC 12 issues (Reported by George Joseph)
- [ASTERISK-29655] - res_pjsip_session: No video to caller if no camera
-                    available (Reported by Michael Auracher)
- [ASTERISK-29638] - res_pjsip_session: No video after early media (Reported by
-                    Michael Auracher)
- [ASTERISK-28518] - chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up
-                    Dahdi Call On Hold (Reported by Josh Alberts)
- [ASTERISK-29990] - chan_dahdi: adding ring cadences is not idempotent on dahdi
-                    restart (Reported by N A)
- [ASTERISK-30007] - chan_iax2: Prevent crashes due to attempted encryption with
-                    missing secrets (Reported by N A)
- [ASTERISK-29728] - menuselect: Disabled by default modules that are enabled
-                    are always recompiled (Reported by N A)
- [ASTERISK-30002] - app_meetme: Don't erroneously set global variables when
-                    channel is NULL (Reported by N A)
- [ASTERISK-22246] - Asterisk's "T" flag is ignored when used with "r" or "R"
-                    flags. (documentation bug) (Reported by Rusty Newton)
- [ASTERISK-26582] - Asterisk seems to ignore the "n" parameter for "disable
-                    console colorization" (Reported by Sebastian Gutierrez)
- [ASTERISK-29994] - chan_dahdi: Round robin array size is too small for max
-                    number of groups (Reported by N A)
- [ASTERISK-29843] - Session timers get removed on UPDATE (Reported by Mark
-                    Petersen)
- [ASTERISK-29943] - file.c: seeking to negative file offset is not prevented
-                    (Reported by N A)
- [ASTERISK-29842] - Do not change 180 Ringing to 183 Progress even if
-                    early_media already enabled (Reported by Mark Petersen)
- [ASTERISK-29955] - chan_sip: SIP route header is missing on UPDATE (Reported
-                    by Mark Petersen)
- [ASTERISK-29948] - iostream: Infinite TCP timeout writing data (Reported by N
-                    A)
- [ASTERISK-29253] - Incorrect bridging on transfer (Reported by Yury Kirsanov)
- [ASTERISK-30024] - Failed to sign STIR/SHAKEN payload with functionality not
-                    enabled (Reported by Claude Diderich)
- [ASTERISK-30006] - res_pjsip: UDP transport does not work when
-                    async_operations is greater than 1 (Reported by Ross Beer)
- [ASTERISK-30021] - ast_variable_list_replace_variable uses variable with new
-                    keyword (Reported by Jasper Hafkenscheid)
- [ASTERISK-30023] - cdr_adaptive_odbc: does not support DATETIME database
-                    columns (Reported by Gregory Massel)
- [ASTERISK-30015] - pjsip / WebRTC: Chrome creating large number of SDP
-                    attributes (Reported by Josh Hogan)
- [ASTERISK-29411] - Crash in pjsip_msg_find_hdr_by_name (Reported by LA)
- [ASTERISK-29535] - Segmentation fault in libasteriskpj.so.2 (Reported by
-                    Daniel Bonazzi)
- [ASTERISK-26719] - pbx: Only up to 127 includes in a dialplan context
-                    (AST_PBX_MAX_STACK - 1) (Reported by Tzafrir Cohen)
- [ASTERISK-29988] - REGRESSION: The build process is requiring xmllint or
-                    xmlstarlet ro be installed when it shouldn't (Reported by
-                    George Joseph)
- [ASTERISK-29986] - build: Asterisk 18.11.0 doesn't compile when wget isn't
-                    available (Reported by Stefan Ruijsenaars)
- [ASTERISK-29895] - chan_iax2: Fix misaligned spacing in iax2 show netstats
-                    printout (Reported by N A)
- [ASTERISK-29048] - chan_iax2: "iax2 show registry" shows host for perceived
-                    (Reported by David Herselman)
- [ASTERISK-26689] - res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting
-                    channel for lack of RTP activity (Reported by Dmitriy Serov)
- [ASTERISK-29929] - res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP
-                    activity in one way sessions (Reported by Boris P. Korzun)
- [ASTERISK-29674] - Adjust for 64bit time_t (Reported by Andre Heider)
- [ASTERISK-29961] - RLS: domain part of 'uri' list attribute mismatch with
-                    SUBSCRIBE request (Reported by Alexei Gradinari)
- [ASTERISK-29960] - ari: Retrieving stored recording can returns wrong file
-                    (Reported by Arix)
- [ASTERISK-29950] - SayNumber can handle '01' to '07', but not '08' or '09'
-                    (Reported by Jim Van Meggelen)
- [ASTERISK-29928] - logging messages truncated when using MUSL runtime
-                    (Reported by Philip Prindeville)
- [ASTERISK-29939] - agi: Fix xmldoc bug with set music (Reported by N A)
- [ASTERISK-28891] - documentation: AGICommand_set+music documentation arguments
-                    displayed incorreclty (Reported by Jonathan Harris)
- [ASTERISK-29924] - res_config_pgsql: omit "unsupported column type 'text'"
-                    error (Reported by Boris P. Korzun)
- [ASTERISK-29923] - docs, LICENSE: pbx.digium.com no longer exists (Reported by
-                    N A)
- [ASTERISK-29904] - RLS: Batched Notifications stop working (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29365] - taskprocessor: Can cause assert at shutdown (Reported by
-                    Joshua C. Colp)
- [ASTERISK-29873] - [patch] Queue Realtime load (Reported by Alexei Gradinari)
- [ASTERISK-18416] - [patch] Realtime queue agents unavailable via AMI before a
-                    call event. (Reported by kwk)
- [ASTERISK-27597] - AMI Queuestatus not working (with realtime queue) (Reported
-                    by cagdas kopuz)
- [ASTERISK-29871] - res_prometheus: Failure to load causes FRACKs (Reported by
-                    Mark Petersen)
- [ASTERISK-29886] - Asterisk AMI sends not-valid XML (Reported by Napadailo
-                    Yaroslav)
- [ASTERISK-29888] - res_pjsip_outbound_authenticator_digest: ABRT attempting to
-                    clean up auth_sess (Reported by George Joseph)
- [ASTERISK-29857] - res_tonedetect: fix logic errors in code (Reported by N A)
- [ASTERISK-29854] - func_frame_drop: fix buffer usage typo (Reported by N A)
- [ASTERISK-29869] - rtp sequence number can skip after DTMF under certain
-                    bridges (Reported by Torrey Searle)
- [ASTERISK-29817] - gethostbyname_r is misdetected on NetBSD and causes a build
-                    failure (Reported by Michał Górny)
- [ASTERISK-29698] - Segfault if sorcery object_lifetime_maximum and
-                    qualify_frequency the same value (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29852] - make_version uses GNU-ism that break git-svn-id parsing on
-                    NetBSD (Reported by Michał Górny)
- [ASTERISK-29850] - ast_get_tid() not implemented for NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29851] - rdtsc is not enabled (stubbed out) on NetBSD (Reported by
-                    Michał Górny)
- [ASTERISK-29818] - Build failure on NetBSD due to hmac function collision
-                    (Reported by Michał Górny)
- [ASTERISK-29856] - res_rtp_asterisk: Invalid comparison creates unreachable
-                    code (Reported by N A)
- [ASTERISK-29867] - configure fails if libsrtp dev files are not installed
-                    (Reported by Sean Bright)
- [ASTERISK-29813] - res_pjsip_session doesn't support multipart message bodies
-                    (Reported by George Joseph)
- [ASTERISK-29858] - Regression: Using external pjproject not working after
-                    "hack" commit (Reported by George Joseph)
- [ASTERISK-29859] - VoiceMailMain() fails when encountering non-numeric
-                    CALLERID(num) (Reported by Mark Murawski)
- [ASTERISK-29847] - pbx_variables: ASTSBINDIR is missing (Reported by N A)
- [ASTERISK-29824] - It's hard to make changes to bundled pjproject (Reported by
-                    George Joseph)
- [ASTERISK-29695] - SAY.CONF wrong logic when converting 24hour time to say 12
-                    hour am/pm (Reported by Vincent Dubois)
- [ASTERISK-29664] - PJSIP processing token with % incorrectly (Reported by Dan
-                    Cropp)
- [ASTERISK-29827] - Support for Nordic language syntax in Queues (Reported by
-                    Mark Petersen)
- [ASTERISK-29515] - app_queue: QueueSummary and QueueStatus events don't exist
-                    in documentation (Reported by Luke Escude)
- [ASTERISK-29746] - tcptls.c: TCP client connect fails due to interrupt
-                    (Reported by Kevin Harwell)
- [ASTERISK-29806] - app_queue: extension state incorrect (Reported by Steve
-                    Davies)
- [ASTERISK-29816] - SAY_DTMF_INTERRUPT channel variable is not honored
-                    (Reported by Sean Bright)
- [ASTERISK-28863] - The ast_rtp_codecs_payloads functions don't preserve order
-                    (Reported by George Joseph)
- [ASTERISK-29320] - res_pjsip_sdp_rtp: Codec preference order of remote is not
-                    correct on unhold (Reported by Ross Beer)
- [ASTERISK-29821] - Deadlock in bridge_channel_internal_join() on local
-                    channels. (Reported by Krzysztof Trempala)
- [ASTERISK-29722] - test_timezone_watch breaks during DST to ST transition
-                    (Reported by Josh Soref)
- [ASTERISK-29804] - bundled_pjproject: sip_inv is missing multipart support in
-                    some cases (Reported by George Joseph)
- [ASTERISK-29794] - ast_coredumper does not delete results when requested and a
-                    specific output dir is set (Reported by Frederic Van Espen)
- [ASTERISK-29803] - pbx_variables: cp4 variables is used uninitialized
-                    (Reported by N A)
- [ASTERISK-29766] - pbx_variables: MSet truncates sets after 24 variables
-                    (Reported by N A)
- [ASTERISK-29772] - chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause
-                    a coredump (Reported by Mark Petersen)
- [ASTERISK-29790] - xmldoc: Dump invalid to XML DTD: XSLT (Reported by
-                    Alexander Traud)
- [ASTERISK-29791] - xmldoc: Dump invalid to XML DTD: ACO Matchfield (Reported
-                    by Alexander Traud)
- [ASTERISK-26991] - documentation: Doxygen site is no longer being updated
-                    (Reported by Joshua C. Colp)
- [ASTERISK-20259] - [patch] Update Doxygen Configuration for make progdocs
-                    (Reported by Andrew Latham)
- [ASTERISK-29785] - res_pjsip_sdp_rtp: Warns on every offered crypto suite
-                    (Reported by Alexander Traud)
- [ASTERISK-28219] - res_ari: Channel create and dial may cause "BUG! Must
-                    supply a channel name.." error (Reported by Anil Gupta)
- [ASTERISK-27406] - Infinite loop when out of ports and rtpstart value is odd
-                    (Reported by Thomas Guebels)
- [ASTERISK-28053] - chan_pjsip: Wrong or missing Q.850 reason in CANCEL
-                    (Reported by Simone Lazzaris)
- [ASTERISK-29761] - res: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29763] - main: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29779] - progdocs: Hidden code sections with syntax errors.
-                    (Reported by Alexander Traud)
- [ASTERISK-29732] - progdocs: Fix grouping for latest Doxygen (Reported by
-                    Alexander Traud)
- [ASTERISK-29771] - Crash occurs when 2 realtime sippeers mysql connections are
-                    configured and we have a schema warning (Reported by Mario
-                    Ban)
- [ASTERISK-29776] - stir/shaken: Requires GNU designator (Reported by Alexander
-                    Traud)
- [ASTERISK-29773] - progdocs: doxyref.h outdated (Reported by Alexander Traud)
- [ASTERISK-29765] - xmldoc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29730] - Segfault in __ao2_ref if refdebug = yes (Reported by Alexei
-                    Gradinari)
- [ASTERISK-29762] - channels: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29748] - bridging: Infinite loop when both Local channel halves in
-                    same bridge (Reported by Joshua C. Colp)
- [ASTERISK-29754] - odbc: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29753] - parking: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29756] - res_ari: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29755] - frame: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29750] - stasis: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29752] - app: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29749] - res_xmpp: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29751] - channel: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29737] - chan_iax2: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29747] - res_pjsip: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29743] - bridges: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29742] - addons: Fix for Doxygen. (Reported by Alexander Traud)
- [ASTERISK-29740] - apps: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29741] - tests: Fix for Doxygen (Reported by Alexander Traud)
- [ASTERISK-29735] - progdocs: Avoid multiple use of section labels (Reported by
-                    Alexander Traud)
- [ASTERISK-29734] - progdocs: Use Doxygen \example correctly (Reported by
-                    Alexander Traud)
- [ASTERISK-29736] - bridge_channel: Fix for Doxygen (Reported by Alexander
-                    Traud)
- [ASTERISK-29733] - progdocs: Avoid name with Doxygen \file (Reported by
-                    Alexander Traud)
- [ASTERISK-29744] - app_morsecode: Fix deadlock (Reported by N A)
- [ASTERISK-29705] - app_read: Fix custom terminator functionality regression
-                    (Reported by N A)
- [ASTERISK-29703] - res_pjsip_callerid: Fix OLI parsing (Reported by N A)
- [ASTERISK-29724] - BuildSystem: In POSIX sh, == in place of = is undefined.
-                    (Reported by Alexander Traud)
- [ASTERISK-28040] - pbx: "dialplan reload" is removing minus symbol from
-                    dynamic hints (Reported by Daniel Zanutti)
- [ASTERISK-29702] - sig_analog: Fix truncated buffer copy (Reported by N A)
- [ASTERISK-29391] - VoiceMail does not cancel recording on rerecord hangup
-                    (Reported by N A)
- [ASTERISK-29709] - res_snmp: Not build on recent Debian distributions.
-                    (Reported by Alexander Traud)
- [ASTERISK-29717] - res_config_sqlite: not removed in makeopts.in (Reported by
-                    Alexander Traud)
- [ASTERISK-29710] - stasis: Clang 13 warns about the unused but set variable
-                    dispatched. (Reported by Alexander Traud)
- [ASTERISK-29711] - aelparse: GCC 11.2 found two maybe uninitialized (Reported
-                    by Alexander Traud)
- [ASTERISK-29713] - GCC 11.2: two stringop-overread (Reported by Alexander
-                    Traud)
- [ASTERISK-29682] - Squash compiler issues generated by gcc 11 (Reported by
-                    George Joseph)
- [ASTERISK-29693] - Using --with-crypto and --with-ssl fails on a recompile
-                    (Reported by George Joseph)
- [ASTERISK-27816] - func_talkdetect's logic is completely broken (Reported by
-                    Moritz Fain)
- [ASTERISK-26497] - make install downloads x86_32 variants of external modules
-                    on non Intel architectures (Reported by Corey Farrell)
- [ASTERISK-29691] - stun: Not all users provide a dst to ast_stun_request
-                    (Reported by Dennis Haney)
- [ASTERISK-20219] - [patch] IAX2 Call Encryption Fails with RSA authentication
-                    (Reported by Michael Munger)
- [ASTERISK-29402] - res_pjsip_t38: Socket is bound to IPv4/IPv6 but platform
-                    does not support it (Reported by Matthew Kern)
- [ASTERISK-29673] - app_read: Fix null pointer crash regression (Reported by N
-                    A)
- [ASTERISK-29671] - res_rtp_asterisk: memory leak (Reported by Jean Aunis -
-                    Prescom)
- [ASTERISK-29668] - ari: Listing bridges fails when dialing bridge exists
-                    (Reported by Joshua C. Colp)
- [ASTERISK-29663] - messaging: AMI MessageSend does not support same parameters
-                    as dialplan application (Reported by Brian J. Murrell)
- [ASTERISK-29578] - app_queue: Custom device state using included hints do not
-                    update (Reported by N A)
- [ASTERISK-29660] - Build failure when disabling PJSIP support (Reported by
-                    Guido Falsi)
- [ASTERISK-29654] - pjproject includes trailing whitespace in sdp format
-                    attributes (Reported by George Joseph)
- [ASTERISK-29635] - MP3Player don' t work with actual mpg123 versions (Reported
-                    by Carlos Oliva)
- [ASTERISK-29629] - ARI external media channel creation doesn't set option data
-                    (Reported by sungtae kim)
- [ASTERISK-27176] - test_abstract_jb: frames leak (Reported by Corey Farrell)
- [ASTERISK-29634] - res_snmp: gcc 11 needs -fPIC to compile correctly (Reported
-                    by George Joseph)
- [ASTERISK-29630] - Asterisk is unable to read extended number format terminfo
-                    files (Reported by Sean Bright)
- [ASTERISK-28004] - dns: Core ast_dns_get_nameservers does not support
-                    configured IPv6 servers (Reported by Isaac McDonald)
- [ASTERISK-29618] - ConfBridge errors on creation conference room (Reported by
-                    Alexander Zharov)
- [ASTERISK-29622] - ARI: external media create doesn't use body parameter
-                    (Reported by sungtae kim)
- [ASTERISK-29614] - app_agent_pool: XML Doc: unterminated entity reference
-                    (Reported by Alexander Traud)
- [ASTERISK-29609] - Subsequent 'ael reload' will cause a lock up (Reported by
-                    Mark Murawski)
- [ASTERISK-28701] - app_queue: Core reload resets queue stats, even when
-                    keepstats=yes (Reported by Luke Escude)
- [ASTERISK-29616] - res_rtp_asterisk: sqrt(.) requires the header math.h.
-                    (Reported by Alexander Traud)
- [ASTERISK-29518] - sig_analog: FCG_CAMA fails to signal ANI spill when using
-                    MF signaling (Reported by Sarah Autumn)
- [ASTERISK-29582] - res_pjproject: Can't map pjproject log messages to Asterisk
-                    TRACE (Reported by George Joseph)
- [ASTERISK-29575] - app_milliwatt: Milliwatt application doesn't use the proper
-                    timings (Reported by N A)
- [ASTERISK-20339] - chan_mgcp, resp_pktccops ast_debug support (Reported by
-                    Tomas Maldonado)
- [ASTERISK-29540] - aelparse: include of context with timings fails (Reported
-                    by Alexander Traud)
- [ASTERISK-29539] - Segmentation fault at ast_writestream() when write handler
-                    not defined (happens with OGG/Speex) (Reported by Ernani
-                    José Camargo Azevedo)
- [ASTERISK-29494] - cdr_adaptive_odbc: Prevent throwing warnings if CDR
-                    filtering is used (Reported by N A)
- [ASTERISK-29513] - statsd: Remove non-standard metric type Meter (Reported by
-                    Rijnhard Hessel)
- [ASTERISK-12] - app_voicemail2 became a bit silent, lately (Reported by siggi)
- [ASTERISK-29526] - G729 audio gets corrupted by Asterisk due to smoother
-                    (Reported by under)
- [ASTERISK-29392] - chan_iax2: Asterisk crashes when queueing video with format
-                    (Reported by Michael Welk)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30178] - extend user_eq_phone behavior to local uri's (Reported by
-                    Michael Bradeen)
- [ASTERISK-30046] - Reimplement res/res_crypto.c internals with EVP_PKEY
-                    interface to Openssl API's (Reported by Philip Prindeville)
- [ASTERISK-30045] - Add test coverage to res/res_crypto.c functionality
-                    (Reported by Philip Prindeville)
- [ASTERISK-30209] - pbx_variables: Use const char for
-                    pbx_substitute_variables_helper_full_location (Reported by N
-                    A)
- [ASTERISK-30185] - res_geolocation: Allow location parameters to be specified
-                    in profiles (Reported by George Joseph)
- [ASTERISK-30177] - res_geolocation: Add option to suppress empty elements
-                    (Reported by George Joseph)
- [ASTERISK-30182] - res_geolocation: Add built-in profiles to use in fully
-                    dynamic configurations (Reported by George Joseph)
- [ASTERISK-29906] - [patch] update RLS to reflect the changes to the lists
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30163] - general: fix minor formatting issues (Reported by N A)
- [ASTERISK-30164] - chan_iax2: Add missing option documentation (Reported by N
-                    A)
- [ASTERISK-30153] - logger: Improve log levels (Reported by N A)
- [ASTERISK-30160] - cdr.conf: Remove obsolete app_mysql reference (Reported by
-                    N A)
- [ASTERISK-30159] - general: Remove obsolete SVN references (Reported by N A)
- [ASTERISK-30128] - Create PJSIP interface module for Geolocation (Reported by
-                    George Joseph)
- [ASTERISK-30127] - Create core Geolocation capability for Asterisk (Reported
-                    by George Joseph)
- [ASTERISK-30089] - general: fix typos (Reported by N A)
- [ASTERISK-30050] - Upgrade Asterisk to bundled pjproject 2.12.1 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30090] - xmldocs: Use example tags for examples (Reported by N A)
- [ASTERISK-29891] - [patch] provide a display name for RLS subscriptions
-                    (Reported by Alexei Gradinari)
- [ASTERISK-30086] - res_parking: Warn when invalid parking space requested
-                    (Reported by N A)
- [ASTERISK-30058] - Evaluate dialplan functions and variables in agi exec
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-30027] - ari: expose channel driver's unique id (i.e. Call-ID for
-                    chan_sip/chan_pjsip) in ARI channel resource (Reported by
-                    Moritz Fain)
- [ASTERISK-29845] - res_pjsip_outbound_registration: Show time remaining until
-                    registration lapses (Reported by N A)
- [ASTERISK-24827] - Missing documentation for chan_dahdi dial string ring
-                    cadences (Reported by Scott Griepentrog)
- [ASTERISK-29940] - general: Add since tags to xmldocs (Reported by N A)
- [ASTERISK-30008] - samples: Remove obsolete config files (Reported by N A)
- [ASTERISK-29726] - Add Asterisk External Application Protocol (AEAP)
-                    implementation (Reported by Kevin Harwell)
- [ASTERISK-29951] - app_mf, app_sf: Return -1 on hangup (Reported by N A)
- [ASTERISK-29954] - app_meetme: Emit warning if conference not found (Reported
-                    by N A)
- [ASTERISK-29935] - build: Remove leftover build references (Reported by N A)
- [ASTERISK-29351] - Qualify pjproject 2.12 for Asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29976] - Should Readme include information about install_prereq
-                    script? (Reported by Marcel Wagner)
- [ASTERISK-29970] - Use pkg-config to find libxml2 headers and libraries
-                    (Reported by Hugh McMaster)
- [ASTERISK-25716] - Documentation: Document explanations and examples for
-                    possible values of DIALSTATUS (Reported by Rusty Newton)
- [ASTERISK-29980] - build: External binary modules don't use https (Reported by
-                    INVADE International Ltd.)
- [ASTERISK-29967] - pbx_builtins: Add missing documentation (Reported by N A)
- [ASTERISK-29909] - app_queue: Add support for withdrawing a call (Reported by
-                    Kfir Itzhak)
- [ASTERISK-29353] - Qualify jansson 2.14 for asterisk (Reported by George
-                    Joseph)
- [ASTERISK-29897] - channels: Increase core debug levels for chatty debugs
-                    (Reported by N A)
- [ASTERISK-29896] - xmldocs: Add since tag (Reported by N A)
- [ASTERISK-29861] - asterisk.h: add macro for curl user agent (Reported by N A)
- [ASTERISK-29809] - curl, stir_shaken: refactor curl code (Reported by N A)
- [ASTERISK-29920] - app_voicemail: Warn if trying to manage nonexistent mailbox
-                    (Reported by N A)
- [ASTERISK-29925] - func_db: Warn about malformed key names (Reported by N A)
- [ASTERISK-29866] - cli: add core dump information to core show settings
-                    (Reported by N A)
- [ASTERISK-29898] - documentation: Add default attributes to documentation
-                    (Reported by N A)
- [ASTERISK-29900] - app_mp3: Document and warn about https incompatibility
-                    (Reported by N A)
- [ASTERISK-29877] - app_mf: Allow reading a maximum number of digits (Reported
-                    by N A)
- [ASTERISK-29832] - Enable pickup on channel after having received 183 Progress
-                    (Reported by Mark Petersen)
- [ASTERISK-29831] - Queue don't play "thank-you" when here is no hold time
-                    announcements (Reported by Mark Petersen)
- [ASTERISK-28890] - res_pjsip_sdp_rtp: Keepalive not supported for video
-                    streams (Reported by Luke Escude)
- [ASTERISK-29855] - frame.h: fix CNG documentation typo (Reported by N A)
- [ASTERISK-29848] - documentation: Document special system and channel
-                    variables (Reported by N A)
- [ASTERISK-29819] - utils.c: Remove all usages of ast_gethostbyname() (Reported
-                    by Sean Bright)
- [ASTERISK-29815] - dsp: Define magic number as macro (Reported by N A)
- [ASTERISK-29807] - cli: add module refresh command (Reported by N A)
- [ASTERISK-29829] - app_mp3: Throw warning if attempting to play a nonexistent
-                    stream (Reported by N A)
- [ASTERISK-24427] - Documentation is missing for a few AMI Events - Including
-                    CDR and events triggered after the QueueStatus action
-                    (Reported by Dafi Ni)
- [ASTERISK-29795] - DIALEDPEERNUMBER not set on destination channel for Queue
-                    calls (Reported by Mark Petersen)
- [ASTERISK-29801] - app.c: Throw warnings for nonexistent options (Reported by
-                    N A)
- [ASTERISK-29797] - Support for Danish language syntax in VM (Reported by Mark
-                    Petersen)
- [ASTERISK-29800] - strings: Fix misusage in comment examples (Reported by N A)
- [ASTERISK-29758] - configs: Minor updates to sample configs (Reported by N A)
- [ASTERISK-29745] - pbx: Add public API for more elegant variable substitution
-                    with extensions (Reported by N A)
- [ASTERISK-29729] - Incompatibility with newer spandsp releases (3.0.0+)
-                    (Reported by Dustin Marquess)
- [ASTERISK-29777] - documentation: Standardize example syntax (Reported by N A)
- [ASTERISK-29715] - app_voicemail: Refactor email generation functions
-                    (Reported by N A)
- [ASTERISK-29727] - Add type for JSON stasis message RTCP Report Received/Sent
-                    (Reported by Boris P. Korzun)
- [ASTERISK-29714] - Spelling errors (Reported by Josh Soref)
- [ASTERISK-29707] - chan_iax2: Allow both key and secret to be specified at
-                    dial time (Reported by N A)
- [ASTERISK-29662] - Add mix option to Playback application for say and filename
-                    (Reported by Shloime Rosenblum)
- [ASTERISK-29637] - Add support for future dates in Say.c (Reported by Shloime
-                    Rosenblum)
- [ASTERISK-29525] - PJSIP remove_existing unavailable contacts (Reported by
-                    Joseph Nadiv)
- [ASTERISK-29661] - func_vmcount: Add support for multiple mailboxes (Reported
-                    by N A)
- [ASTERISK-29275] - Support of MIME-type for wav16 (Reported by Boris P.
-                    Korzun)
- [ASTERISK-29529] - Add custom logging level (Reported by N A)
- [ASTERISK-29472] - res_pjsip: OLI/ANI2 support missing (Reported by N A)
- [ASTERISK-29626] - app_stack: Include calling location if attempting to branch
-                    to nonexistent location (Reported by N A)
- [ASTERISK-29632] - Add option to Application_VoiceMail to suppress
-                    instructions only when a custom greeting is present (Reported
-                    by Charlie Smurthwaite)
- [ASTERISK-29605] - chan_iax2: Add ANI2 (Reported by N A)
- [ASTERISK-29508] - STUN server address refresh (Reported by Sébastien Duthil)
- [ASTERISK-29612] - bridge_basic: Don't throw warning if attended transfer is
-                    cancelled (Reported by N A)
- [ASTERISK-29544] - Media Cache - Delayed remote sound file retrieve delays all
-                    playbacks (Reported by Andre Barbosa)
- [ASTERISK-29495] - Return integer instead of float if response is a whole
-                    number (Reported by N A)
- [ASTERISK-29541] - app_morsecode: Add American Morse code (Reported by N A)
- [ASTERISK-29543] - app_originate: Allow specifying codec(s) to use (Reported
-                    by N A)
- [ASTERISK-29528] - Add support for multiple files for agent announcements
-                    (Reported by N A)
- [ASTERISK-29527] - res_http_media_cache: Cleanup audio format lookup in HTTP
-                    requests (Reported by Sean Bright)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
</package>
<package pkgid="59ab8c3d5b123e12c8a6632a9ad7ec25f85900c4852d606ed0a6d712999523b0" name="asterisk" arch="src">
  <version epoch="0" ver="21.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.0.1-1" date="1669896000">- The Asterisk Development Team would like to announce the release of Asterisk
- 16.29.1, 18.15.1, 19.7.1, and 20.0.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
- The release of Asterisk 16.29.1, 18.15.1, 19.7.1, and 20.0.1 resolves issues
- reported by the community and would have not been possible without your
- participation. Thank you!
-
- The following issue is resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30103] chan_ooh323 vulnerability in calling/called party IE
-                  (Reported By: Michael Bradeen)
- [ASTERISK-30176] GetConfig can read files outside of Asterisk (Reported By:
-                  shawty)
- [ASTERISK-30244] Occasional crash when TCP/TLS connection terminated and
-                  subscription persistence is removed (Reported By: nappsoft)
- [ASTERISK-30338] Backport 2.13 security fixes from pjproject
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.29.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.15.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-19.7.1
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.0.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce 
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community 
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed. 
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
</package>
<package pkgid="e2eebd9532a89876b1f8aba52f667ba6d06c12eac69a4cd9584e2a8509e48a71" name="asterisk" arch="src">
  <version epoch="0" ver="21.0.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.1.0-1" date="1673524800">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.1.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.1.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Security bugs fixed in this release:
- ------------------------------------
-
- [ASTERISK-30338] - pjproject: Backport security fixes from 2.13 (Reported by
-                    Benjamin Keith Ford)
- [ASTERISK-30176] - manager: GetConfig can read files outside of Asterisk
-                    (Reported by shawty)
- [ASTERISK-30103] - chan_ooh323 Vulnerability in calling/called party IE
-                    (Reported by Michael Bradeen)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30328] - Typo in from_domain description on res_pjsip
-                    configuration documentation (Reported by Marcel Wagner)
- [ASTERISK-30316] - res_pjsip: Documentation should point out default if
-                    contact_user is not being set for outbound registrations
-                    (Reported by Marcel Wagner)
- [ASTERISK-30289] - xmldoc: Allow XML docs to be reloaded (Reported by N A)
- [ASTERISK-30327] - rtp_engine.h: Remove obsolete example usage (Reported by
-                    N A)
- [ASTERISK-30286] - app_mixmonitor: Add option to use real Caller ID for
-                    Caller ID (Reported by N A)
- [ASTERISK-30308] - pbx_builtins: Allow Answer to return immediately
-                    (Reported by N A)
- [ASTERISK-30295] - test_json: Remove duplicated static function (Reported by
-                    N A)
- [ASTERISK-30290] - file.c: Don't emit warnings on winks. (Reported by N A)
- [ASTERISK-30241] - res_pjsip_gelocation: Downgrade some NOTICE scope trace
-                    debugs to DEBUG level (Reported by N A)
- [ASTERISK-30223] - features: add no-answer option to Bridge application
-                    (Reported by N A)
- [ASTERISK-30158] - PJSIP: Add new 100rel option "peer_supported" (Reported by
-                    Maximilian Fridrich)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30349] - app_if: Format truncation error (Reported by George
-                    Joseph)
- [ASTERISK-30344] - ari: Memory leak in create when specifying JSON (Reported
-                    by Saken)
- [ASTERISK-30283] - app_voicemail: Fix msg_create_from_file not sending email
-                    to user (Reported by N A)
- [ASTERISK-30265] - res_pjsip_session: Fix missing PLAR support on INVITEs
-                    (Reported by N A)
- [ASTERISK-29793] - adsi: CAS is malformed (Reported by N A)
- [ASTERISK-30311] - func_presencestate: Fix invalid memory access. (Reported
-                    by N A)
- [ASTERISK-30336] - sig_analog: Fix no timeout duration (Reported by N A)
- [ASTERISK-30244] - res_pjsip_pubsub: Occasional crash when TCP/TLS
-                    connection terminated and subscription persistence is removed
-                    (Reported by nappsoft)
- [ASTERISK-30184] - res_pjsip_session: re-INVITE after answering results in
-                    wrong stream direction of first call leg (Reported by
-                    Maximilian Fridrich)
- [ASTERISK-29998] - sla: deadlock when calling SLAStation application
-                    (Reported by N A)
- [ASTERISK-30321] - Build: Embedded blobs have executable stacks (Reported by
-                    George Joseph)
- [ASTERISK-30293] - Memory leak in JSON_DECODE (Reported by David Uczen)
- [ASTERISK-30314] - res_agi: RECORD FILE doesn't respect "transmit_silence"
-                    asterisk.conf option (Reported by Joshua C. Colp)
- [ASTERISK-30285] - manager.c: Remove outdated documentation (Reported by N
-                    A)
- [ASTERISK-30282] - CI: Coredump output isn't saved when running unittests
-                    (Reported by George Joseph)
- [ASTERISK-30076] - app_stack: Incorrect exit location in predial handlers
-                    logged (Reported by N A)
- [ASTERISK-30281] - chan_rtp: Local address being used before being set
-                    (Reported by George Joseph)
- [ASTERISK-28689] - res_pjsip: Crash when locking group lock when sending
-                    stateful response (Reported by Jesse Ross)
- [ASTERISK-30278] - tcptls: Abort occurs if SSL error is logged if
-                    MALLOC_DEBUG is enabled (Reported by N A)
- [ASTERISK-30217] - Registration do not allow multiple proxies (Reported by
-                    Igor Goncharovsky)
- [ASTERISK-30273] - test_mwi: compilation fails on 32-bit Debian (Reported by
-                    N A)
- [ASTERISK-30193] - chan_pjsip should return all codecs on a re-INVITE
-                    without SDP (Reported by Henning Westerholt)
- [ASTERISK-30258] - Dialing API: Cancel a running async thread, does not
-                    always cancel all calls (Reported by Frederic LE FOLL)
- [ASTERISK-30274] - chan_dahdi: Unavailable channels are BUSY (Reported by N
-                    A)
- [ASTERISK-30264] - res_pjsip: Subscription handlers do not get cleanly
-                    unregistered, causing crash (Reported by N A)
- [ASTERISK-30248] - ast_get_digit_str adds bogus initial delimiter if first
-                    character not to be spoken (Reported by David Woolley)
- [ASTERISK-30213] - Make crypto_load() reentrant and handle symlinks
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30256] - chan_dahdi: Fix format truncation warnings (Reported by N
-                    A)
- [ASTERISK-30239] - Prometheus plugin crashes Asterisk when using local
-                    channel (Reported by Joeran Vinzens)
- [ASTERISK-30237] - res_prometheus: Crash when scraping bridges (Reported by
-                    Igor Yeroshev)
- [ASTERISK-30245] - db: ListItems is incorrect (Reported by N A)
- [ASTERISK-30243] - func_logic: IF function complains if both branches are
-                    empty (Reported by N A)
- [ASTERISK-30232] - Initialize stack-based ast_test_capture structures
-                    correctly (Reported by Philip Prindeville)
- [ASTERISK-30220] - func_scramble: Fix segfault due to null pointer deref
-                    (Reported by N A)
- [ASTERISK-30235] - res_crypto and tests: Memory issues and and uninitialized
-                    variable error (Reported by George Joseph)
- [ASTERISK-30234] - res_geolocation: may be used uninitialized error in
-                    geoloc_config.c (Reported by George Joseph)
- [ASTERISK-30226] - REGRESSION: res_crypto complains about the stir_shaken
-                    directory in /var/lib/asterisk/keys (Reported by George
-                    Joseph)
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-21502] - New SIP Channel Driver: add Advice of Charge support
-                    (Reported by Matt Jordan)
- [ASTERISK-30322] - res_hep: Add capture agent name support (Reported by N A)
- [ASTERISK-29497] - Add conditional branch applications (Reported by N A)
- [ASTERISK-30150] - res_pjsip_session: Add support for custom parameters
-                    (Reported by N A)
- [ASTERISK-30305] - chan_dahdi: Allow FXO channels to start immediately
-                    (Reported by N A)
- [ASTERISK-30284] - app_mixmonitor: Add option to delete recording file when
-                    done (Reported by N A)
- [ASTERISK-30263] - res_pjsip_notify: Allow using pjsip_notify.conf from AMI
-                    (Reported by N A)
- [ASTERISK-30146] - res_pjsip_logger: Add method-based log filtering
-                    (Reported by N A)
- [ASTERISK-30091] - cdr: Allow CDRs to ignore call state changes (Reported by N
-                    A)
- [ASTERISK-30254] - res_tonedetect: Add audible ringback detection to
-                    TONE_DETECT (Reported by N A)
- [ASTERISK-30032] - Support of mediasec SIP headers and SDP attributes
-                    (Reported by Maximilian Fridrich)
- [ASTERISK-30216] - app_bridgewait: Add option for BridgeWait to not answer
-                    (Reported by N A)
- [ASTERISK-30179] - app_amd: Allow audio to be played while AMD is running
-                    (Reported by N A)
- [ASTERISK-29432] - New function to allow access to any channel (Reported by
-                    N A)
- [ASTERISK-30222] - func_strings: Add trim functions (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.1.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
</package>
<package pkgid="bffa09d1c8a9dba3003186e46810cd9e9834f3438f9ff7f157715e0c055b2038" name="asterisk" arch="src">
  <version epoch="0" ver="21.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.0-1" date="1678363200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.0.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.0 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- New Features made in this release:
- ----------------------------------
-
- [ASTERISK-29810] - app_signal: Add channel signaling applications (Reported by
-                    N A)
- [ASTERISK-30262] - res_pjsip_session: Allow a context to be specified for
-                    overlap dialing (Reported by N A)
- [ASTERISK-30319] - Add BYE Reason support for SIP (Reported by Igor
-                    Goncharovsky)
- [ASTERISK-30180] - app_broadcast: Add a channel audio multicasting application
-                    (Reported by N A)
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-27830] - Asterisk crashes on Invalid UTF-8 string (Reported by
-                    AvayaXAsterisk)
- [ASTERISK-30354] - chan_iax2: Lack of formats prior to receiving voice frames
-                    causes jitterbuffer to stall (Reported by N A)
- [ASTERISK-30162] - when chan_iax is used to relay calls, no ringing indication
-                    is played (Reported by Jaco Kroon)
- [ASTERISK-30424] - pjproject_bundled: cross-compilation broken when ssl
-                    autodetected (Reported by Nick French)
- [ASTERISK-30388] - res_phoneprov: Stale SERVER variable when multi-homed
-                    (Reported by cmaj)
- [ASTERISK-30419] - pjsip: Crash when sending NOTIFY in PJSIP 2.13 (Reported by
-                    Ross Beer)
- [ASTERISK-30417] - Copy/Paste error in UnpauseQueueMember (Reported by Sean
-                    Bright)
- [ASTERISK-29604] - ari: Segfault with lots of calls (Reported by Danila
-                    Evgrafov)
- [ASTERISK-30406] - pbx_ael: Global variables are not expanded (Reported by
-                    Sean Bright)
- [ASTERISK-30391] - res_rtp_asterisk: Issue with transcoding g722 after MES
-                    changes (Reported by George Joseph)
- [ASTERISK-30345] - loader.c: Modules that decline to load cannot be reloaded
-                    (Reported by N A)
- [ASTERISK-30351] - manager: Originate variables are not added when setvar used
-                    in manager.conf (Reported by Sebastian Gutierrez)
- [ASTERISK-30369] - res_pjsip: Websockets from same IP shut down when they
-                    shouldn't be (Reported by Joshua C. Colp)
- [ASTERISK-30379] - http: fix NULL pointer dereference while enable_status on
-                    TLS-only (Reported by Boris P. Korzun)
- [ASTERISK-30375] - res_http_media_cache: Crash when URL has no path component
-                    (Reported by Sean Bright)
- [ASTERISK-30367] - pbx: Fix outdated channel snapshots with pbx_exec (Reported
-                    by N A)
- [ASTERISK-28767] - chan_pjsip: Caller ID not used when checking for extension,
-                    callerid supplement executed too late (Reported by Oleg)
- [ASTERISK-30350] - res_pjsip_sdp_rtp: rtp_timeout_hold is not used when
-                    moh_passthrough has call on hold (Reported by Benjamin Keith
-                    Ford)
- [ASTERISK-30240] - app voicemail odbc build error with gcc 11.1 (Reported by
-                    Michael Bradeen)
- [ASTERISK-30100] - res_pjsip: Path is ignored on INVITE to endpoint (Reported
-                    by Yury Kirsanov)
- [ASTERISK-30198] - Error `Too many open files` occurs after about ~8000 calls
-                    when using mixmonitor (Reported by Julien Alie)
- [ASTERISK-30347] - xmldocs: Remove references to removed applications
-                    (Reported by N A)
-
- Improvements made in this release:
- ----------------------------------
-
- [ASTERISK-30411] - app_read: add option to include terminating digit on empty,
-                    terminated strings (Reported by Michael Bradeen)
- [ASTERISK-30405] - app_directory: Add 's' option to skip channel call
-                    (Reported by Michael Bradeen)
- [ASTERISK-30422] - app_senddtmf: add the option for senddtmf to answer
-                    (Reported by Michael Bradeen)
- [ASTERISK-30325] - Upgrade Asterisk to bundled pjproject 2.13 (Reported by
-                    Stanislav Abramenkov)
- [ASTERISK-30404] - app_directory: Add reading directory configuration from
-                    custom file (Reported by Michael Bradeen)
- [ASTERISK-29913] - func_json: Adds multi-level and array parsing to
-                    JSON_DECODE (Reported by N A)
- [ASTERISK-30353] - func_frame_trace: Print text for text frames (Reported by N
-                    A)
- [ASTERISK-30361] - json.h: Add missing ast_json_object_real_get (Reported by N
-                    A)
- [ASTERISK-30280] - Create capability to assign a Media Experience Score to RTP
-                    streams (Reported by George Joseph)
- [ASTERISK-30332] - func_callerid: Warn if invalid redirecting reason provided
-                    (Reported by N A)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.0
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints:  Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals:  Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml:  Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc:  Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
</package>
<package pkgid="7d3ede8fbf95182f9a68f7c423fb5dad5d2dd336d9f5a972bfbb55c1e1601b7d" name="asterisk" arch="src">
  <version epoch="0" ver="21.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.2.1-1" date="1680523200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.2.1.
- This release is available for immediate download at
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The release of Asterisk 20.2.1 resolves several issues reported by the
- community and would have not been possible without your participation.
-
- Thank you!
-
- The following issues are resolved in this release:
-
- Bugs fixed in this release:
- ---------------------------
-
- [ASTERISK-30469] - res_pjsip_pubsub: Regression for subscription shutdowns
-                    (Reported by N A)
- [ASTERISK-30472] - pbx_ael: Literal usage for variables broken (Reported by
-                    isrl)
-
- For a full list of changes in this release, please see the ChangeLog:
- https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-20.2.1
-
- Thank you for your continued support of Asterisk!</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
</package>
<package pkgid="9e75e2941f1444fe85f465311f179c9905756d120b3b1c2772768d09afc94594" name="asterisk" arch="src">
  <version epoch="0" ver="21.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.0-1" date="1684843200">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release 20.3.0
- =============================
-
- Summary:
- --------
-
- Set up new ChangeLogs directory
- .github: Add AsteriskReleaser
- chan_pjsip: also return all codecs on empty re-INVITE for late offers
- cel: add local optimization begin event
- core: Cleanup gerrit and JIRA references. (#57)
- .github: Fix CherryPickTest to only run when it should
- .github: Fix reference to CHERRY_PICK_TESTING_IN_PROGRESS
- .github: Remove separate set labels step from new PR
- .github: Refactor CP progress and add new PR test progress
- res_pjsip: mediasec: Add Security-Client headers after 401
- .github: Add cherry-pick test progress labels
- LICENSE: Update link to trademark policy.
- chan_dahdi: Add dialmode option for FXS lines.
- .github: Update issue templates
- .github: Remove unnecessary parameter in CherryPickTest
- Initial GitHub PRs
- Initial GitHub Issue Templates
- pbx_dundi: Fix PJSIP endpoint configuration check.
- Revert "app_queue: periodic announcement configurable start time."
- res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
- pbx_dundi: Add PJSIP support.
- install_prereq: Add Linux Mint support.
- chan_pjsip: fix music on hold continues after INVITE with replaces
- voicemail.conf: Fix incorrect comment about #include.
- app_queue: Fix minor xmldoc duplication and vagueness.
- test.c: Fix counting of tests and add 2 new tests
- res_calendar: output busy state as part of show calendar.
- loader.c: Minor module key check simplification.
- ael: Regenerate lexers and parsers.
- bridge_builtin_features: add beep via touch variable
- res_mixmonitor: MixMonitorMute by MixMonitor ID
- format_sln: add .slin as supported file extension
- res_agi: RECORD FILE plays 2 beeps.
- func_json: Fix JSON parsing issues.
- app_senddtmf: Add SendFlash AMI action.
- app_dial: Fix DTMF not relayed to caller on unanswered calls.
- configure: fix detection of re-entrant resolver functions
- cli: increase channel column width
- app_queue: periodic announcement configurable start time.
- make_version: Strip svn stuff and suppress ref HEAD errors
- res_http_media_cache: Introduce options and customize
- main/iostream.c: fix build with libressl
- contrib: rc.archlinux.asterisk uses invalid redirect.
-
- User Notes:
- -----------
-
- ### cel: add local optimization begin event
-     The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used by itself or in conert
-     with the existing AST_CEL_LOCAL_OPTIMIZE to book-end local channel
-     optimizaion.
-
- ### chan_dahdi: Add dialmode option for FXS lines.
-     A "dialmode" option has been added which allows specifying, on a
-     per-channel basis, what methods of subscriber dialing (pulse and/or tone)
-     are permitted. Additionally, this can be changed on a channel at any point
-     during a call using the CHANNEL function.
-
- ### pbx_dundi: Add PJSIP support.
-     DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-     pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- ### cli: increase channel column width
-     This change increases the display width on 'core show channels' amd 'core
-     show channels verbose' For 'core show channels', the Channel name field is
-     increased to 64 characters and the Location name field is increased to 32
-     characters. For 'core show channels verbose', the Channel name field is
-     increased to 80 characters, the Context is increased to 24 characters and
-     the Extension is increased to 24 characters.
-
- ### app_senddtmf: Add SendFlash AMI action.
-     The SendFlash AMI action now allows sending a hook flash event on a
-     channel.
-
- ### res_http_media_cache: Introduce options and customize
-     The res_http_media_cache module now attempts to load configuration from the
-     res_http_media_cache.conf file. The following options were added:
-      * timeout_secs
-      * user_agent
-      * follow_location
-      * max_redirects
-      * protocols
-      * redirect_protocols
-      * dns_cache_timeout_secs
-
- ### test.c: Fix counting of tests and add 2 new tests
-     The "tests" attribute of the "testsuite" element in the output XML now
-     reflects only the tests actually requested to be executed instead of all
-     the tests registered. The "failures" attribute was added to the "testsuite"
-     element. Also added two new unit tests that just pass and fail to be used
-     for testing CI itself.
-
- ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-     It is now possible to specify the MixMonitorID when calling the manager
-     action: MixMonitorMute. This will allow an individual MixMonitor instance
-     to be muted via ID. The MixMonitorID can be stored as a channel variable
-     using the 'i' MixMonitor option and is returned upon creation if this
-     option is used. As part of this change, if no MixMonitorID is specified in
-     the manager action MixMonitorMute, Asterisk will set the mute flag on all
-     MixMonitor audiohooks on the channel. Previous behavior would set the flag
-     on the first MixMonitor audiohook found.
-
- ### bridge_builtin_features: add beep via touch variable
-     Add optional touch variable: TOUCH_MIXMONITOR_BEEP(interval) Setting
-     TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in seconds
-     will result in a periodic beep being played to the monitored channel upon
-     MixMontior/Monitor feature start. If an interval less than 5 seconds is
-     specified, the interval will default to 5 seconds. If the value is set to
-     an invalid interval, the default of 15 seconds will be used.
-
- ### format_sln: add .slin as supported file extension
-     format_sln now recognizes '.slin' as a valid file extension in addition to
-     the existing '.sln' and '.raw'.
-
- Upgrade Notes:
- --------------
-
- ### cel: add local optimization begin event
-     The existing AST_CEL_LOCAL_OPTIMIZE can continue to be used as-is and the
-     AST_CEL_LOCAL_OPTIMIZE_BEGIN event can be ignored if desired.
-
- Closed Issues:
- --------------
-
- #35: [New Feature]: chan_dahdi: Allow disabling pulse or tone dialing
- #39: [Bug]: Remove .gitreview from repository.
- #43: [Bug]: Link to trademark policy is no longer correct
- #48: [bug]: res_pjsip: Mediasec requires different headers on 401 response
- #52: [improvement]: Add local optimization begin cel event
-
- ### For more details, see:
- https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.0.md</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
</package>
<package pkgid="d3bbb938049287cdb24162e69c6264ee49e686b63c68b84d75be335a894bd655" name="asterisk" arch="src">
  <version epoch="0" ver="21.3.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.3.1-1" date="1688731200">- The Asterisk Development Team would like to announce security release Asterisk
- 20.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.3.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- https://github.com/asterisk/asterisk/security/advisories/GHSA-4xjp-22g4-9fxm
-
-
- Change Log for Release 20.3.1
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.0...20.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- apply_patches: Use globbing instead of file/sort.
- apply_patches: Sort patch list before applying
- pjsip: Upgrade bundled version to pjproject 2.13.1
-
- User Notes:
- -----------
-
- - ### res_http_media_cache: Introduce options and customize
-   The res_http_media_cache module now attempts to load configuration from the
-   res_http_media_cache.conf file.
-   The following options were added:
-     * timeout_secs
-     * user_agent
-     * follow_location
-     * max_redirects
-     * protocols
-     * redirect_protocols
-     * dns_cache_timeout_secs
-
- - ### format_sln: add .slin as supported file extension
-   format_sln now recognizes '.slin' as a valid file extension in addition to
-   the existing '.sln' and '.raw'.
-
- - ### bridge_builtin_features: add beep via touch variable
-   Add optional touch variable : TOUCH_MIXMONITOR_BEEP(interval)
-   Setting TOUCH_MIXMONITOR_BEEP/TOUCH_MONITOR_BEEP to a valid interval in
-   seconds will result in a periodic beep being played to the monitored channel
-   upon MixMontior/Monitor feature start.
-   If an interval less than 5 seconds is specified, the interval will default
-   to 5 seconds. If the value is set to an invalid interval, the default of 15
-   seconds will be used.
-
- - ### app_senddtmf: Add SendFlash AMI action.
-   The SendFlash AMI action now allows sending a hook flash event on a channel.
-
- - ### res_mixmonitor: MixMonitorMute by MixMonitor ID
-   It is now possible to specify the MixMonitorID when calling the manager
-   action: MixMonitorMute.  This will allow an individual MixMonitor instance to
-   be muted via ID.
-   The MixMonitorID can be stored as a channel variable using the 'i'
-   MixMonitor option and is returned upon creation if this option is used.
-   As part of this change, if no MixMonitorID is specified in the manager
-   action MixMonitorMute, Asterisk will set the mute flag on all MixMonitor
-   audiohooks on the channel. Previous behavior would set the flag on the first
-   MixMonitor audiohook found.
-
- - ### pbx_dundi: Add PJSIP support.
-   DUNDi now supports chan_pjsip. Outgoing calls using PJSIP require the
-   pjsip_outgoing_endpoint option to be set in dundi.conf.
-
- - ### test.c: Fix counting of tests and add 2 new tests
-   The "tests" attribute of the "testsuite" element in the output XML now
-   reflects only the tests actually requested to be executed instead of all the
-   tests registered.
-   The "failures" attribute was added to the "testsuite" element.
-   Also added two new unit tests that just pass and fail to be used for testing
-   CI itself.
-
- - ### cli: increase channel column width
-   This change increases the display width on 'core show channels' amd 'core
-   show channels verbose'.
-   For 'core show channels', the Channel name field is increased to 64
-   characters and the Location name field is increased to 32 characters.
-   For 'core show channels verbose', the Channel name field is increased to 80
-   characters, the Context is increased to 24 characters and the Extension is
-   increased to 24 characters.
-
-
- Upgrade Notes:
- --------------
-
-
- Closed Issues:
- --------------
-
- #193: [bug]: third-party/apply-patches doesn't sort the patch file list before
-              applying</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al.  Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
</package>
<package pkgid="88e1827a51cfc19a8d0684b6cede379481be03652144060174cf5582c2403cab" name="asterisk" arch="src">
  <version epoch="0" ver="21.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.4.0-1" date="1689854400">- The Asterisk Development Team would like to announce the release of Asterisk
- 20.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
-
- Change Log for Release 20.4.0
- =============================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.3.1...20.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- app.h: Move declaration of ast_getdata_result before its first use
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging
- .github: Updates for AsteriskReleaser
- app_voicemail: fix imap compilation errors
- res_musiconhold: avoid moh state access on unlocked chan
- utils: add lock timestamps for DEBUG_THREADS
- .github: Back out triggering PROpenedOrUpdated by label
- .github: Move publish docs to new file CreateDocs.yml
- rest-api: Updates for new documentation site
- .github: Remove result check from PROpenUpdateGateTests
- .github: Fix use of 'contains'
- .github: Add recheck label test to additional jobs
- .github: Fix recheck label typos
- .github: Fix recheck label manipulation
- .github: Allow PR submit checks to be re-run by label
- app_voicemail_imap: Fix message count when IMAP server is unavailable
- res_pjsip_rfc3326: Prefer Q.850 cause code over SIP.
- res_pjsip_session: Added new function calls to avoid ABI issues.
- app_queue: Add force_longest_waiting_caller option.
- pjsip_transport_events.c: Use %zu printf specifier for size_t.
- res_crypto.c: Gracefully handle potential key filename truncation.
- configure: Remove obsolete and deprecated constructs.
- res_fax_spandsp.c: Clean up a spaces/tabs issue
- ast-db-manage: Synchronize revisions between comments and code.
- test_statis_endpoints: Fix channel_messages test again
- res_crypto.c: Avoid using the non-portable ALLPERMS macro.
- tcptls: when disabling a server port, we should set the accept_fd to -1.
- AMI: Add parking position parameter to Park action
- test_stasis_endpoints.c: Make channel_messages more stable
- build: Fix a few gcc 13 issues
- .github: Rework for merge approval
- ast-db-manage: Fix alembic branching error caused by #122.
- app_followme: fix issue with enable_callee_prompt=no (#88)
- sounds: Update download URL to use HTTPS.
- configure: Makefile downloader enable follow redirects.
- res_musiconhold: Add option to loop last file.
- chan_dahdi: Fix Caller ID presentation for FXO ports.
- AMI: Add CoreShowChannelMap action.
- sig_analog: Add fuller Caller ID support.
- res_stasis.c: Add new type 'sdp_label' for bridge creation.
- app_queue: Preserve reason for realtime queues
- .github: Fix issues with cherry-pick-reminder
- indications: logging changes
- .github Ignore error when adding reviewrs to PR
- .github: Update field descriptions for AsteriskReleaser
- callerid: Allow specifying timezone for date/time.
- logrotate: Fix duplicate log entries.
- chan_pjsip: Allow topology/session refreshes in early media state
- chan_dahdi: Fix broken hidecallerid setting.
- .github: Change title of AsteriskReleaser job
- asterisk.c: Fix option warning for remote console.
- .github: Don't add cherry-pick reminder if it's already present
- .github: Fix quoting in PROpenedOrUpdated
- .github: Add cherry-pick reminder to new PRs
- configure: fix test code to match gethostbyname_r prototype.
- res_pjsip_pubsub.c: Use pjsip version for pending NOTIFY check. (#77)
- res_sorcery_memory_cache.c: Fix memory leak
- xml.c: Process XML Inclusions recursively.
- .github: Tweak improvement issue type language.
- .github: Tweak new feature language, and move feature requests elsewhere.
- .github: Fix staleness check to only run on certain labels.
-
- User Notes:
- -----------
-
- ### AMI: Add parking position parameter to Park action
-     New ParkingSpace parameter has been added to AMI action Park.
-
- ### res_musiconhold: Add option to loop last file.
-     The loop_last option in musiconhold.conf now allows the last file in the
-     directory to be looped once reached.
-
- ### AMI: Add CoreShowChannelMap action.
-     New AMI action CoreShowChannelMap has been added.
-
- ### sig_analog: Add fuller Caller ID support.
-     Additional Caller ID properties are now supported on incoming calls to FXS
-     stations, namely the redirecting reason and call qualifier.
-
- ### res_stasis.c: Add new type 'sdp_label' for bridge creation.
-     When creating a bridge using the ARI the 'type' argument now accepts a new
-     value 'sdp_label' which will configure the bridge to add labels for each
-     stream in the SDP with the corresponding channel id.
-
- ### app_queue: Preserve reason for realtime queues
-     Make paused reason in realtime queues persist an Asterisk restart. This
-     was fixed for non-realtime queues in ASTERISK_25732.
-
-
- Upgrade Notes:
- --------------
-
- ### app_queue: Preserve reason for realtime queues
-     Add a new column to the queue_member table: reason_paused VARCHAR(80) so
-     the reason can be preserved.
-
-
- Closed Issues:
- --------------
-
- #45: [bug]: Non-bundled PJSIP check for evsub pending NOTIFY check is
-             insufficient/ineffective
- #55: [bug]: res_sorcery_memory_cache: Memory leak when calling
-             sorcery_memory_cache_open
- #64: [bug]: app_voicemail_imap wrong behavior when losing IMAP connection
- #65: [bug]: heap overflow by default at startup
- #66: [improvement]: Fix preserve reason of pause when Asterisk is restared for
-                     realtime queues
- #73: [new-feature]: pjsip: Allow topology/session refreshes in early media
-                     state
- #87: [bug]: app_followme: Setting enable_callee_prompt=no breaks timeout
- #89: [improvement]: indications: logging changes
- #91: [improvement]: Add parameter on ARI bridge create to allow it to send SDP
-                     labels
- #94: [new-feature]: sig_analog: Add full Caller ID support for incoming calls
- #96: [bug]: make install-logrotate causes logrotate to fail on service restart
- #98: [new-feature]: callerid: Allow timezone to be specified at runtime
- #100: [bug]: sig_analog: hidecallerid setting is broken
- #102: [bug]: Strange warning - 'T' option is not compatible with remote
-              console mode and has no effect.
- #104: [improvement]: Add AMI action to get a list of connected channels
- #108: [new-feature]: fair handling of calls in multi-queue scenarios
- #110: [improvement]: utils - add lock timing information with DEBUG_THREADS
- #116: [bug]: SIP Reason: "Call completed elsewhere" no longer propagating
- #120: [bug]: chan_dahdi: Fix broken presentation for FXO caller ID
- #122: [new-feature]: res_musiconhold: Add looplast option
- #133: [bug]: unlock channel after moh state access
- #136: [bug]: Makefile downloader does not follow redirects.
- #145: [bug]: ABI issue with pjproject and pjsip_inv_session
- #155: [bug]: GCC 13 is catching a few new trivial issues
- #158: [bug]: test_stasis_endpoints.c: Unit test channel_messages is unstable
- #174: [bug]: app_voicemail imap compile errors
- #200: [bug]: Regression: In app.h an enum is used before its declaration.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at 
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md) 
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0) 
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz) 
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) 
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk                             
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"                                 
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits                           
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log             
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
</package>
<package pkgid="92f73cdafee3e432fbf1ffb9619c3f198509043918f10c84d90e6a19a2a2a6bf" name="asterisk" arch="src">
  <version epoch="0" ver="21.4.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 20.5.0-1" date="1697630400">- The Asterisk Development Team would like to announce the release of
- asterisk-20.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/20.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-20.5.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.4.0...20.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- app_macro: Fix locking around datastore access
- Revert "app_stack: Print proper exit location for PBXless channels."
- .github: Use generic releaser
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- core/ari/pjsip: Add refer mechanism
- chan_dahdi: Allow autoreoriginating after hangup.
- audiohook: Unlock channel in mute if no audiohooks present.
- sig_analog: Allow three-way flash to time out to silence.
- res_prometheus: Do not generate broken metrics
- res_pjsip: Enable TLS v1.3 if present.
- func_cut: Add example to documentation.
- extensions.conf.sample: Remove reference to missing context.
- func_export: Use correct function argument as variable name.
- app_queue: Add support for applying caller priority change immediately.
- .github: Fix cherry-pick reminder issues
- chan_iax2.c: Avoid crash with IAX2 switch support.
- res_geolocation: Ensure required 'location_info' is present.
- Adds manager actions to allow move/remove/forward individual messages in a
- particular mailbox folder. The forward command can be used to copy a message
- within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
- required to retrieve message ID's.
- app_voicemail: add CLI commands for message manipulation
- res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp-&gt;themssrc_valid`
- into the scope of the rtp_instance lock.
- .github: Minor tweak to Asterisk Releaser
- .github: Suppress cherry-pick reminder for some situations
- sig_analog: Allow immediate fake ring to be suppressed.
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station user to go on hook when
-     receiving an incoming call and resume from another phone on the same line by
-     going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### core/ari/pjsip: Add refer mechanism
-     There is a new ARI endpoint `/endpoints/refer` for referring an endpoint
-     to some URI or endpoint.
-
- ### chan_dahdi: Allow autoreoriginating after hangup.
-     The autoreoriginate setting now allows for kewlstart FXS channels to
-     automatically reoriginate and provide dial tone to the user again after all
-     calls on the line have cleared. This saves users from having to manually hang
-     up and pick up the receiver again before making another call.
-
- ### sig_analog: Allow three-way flash to time out to silence.
-     The threewaysilenthold option now allows the three-way dial tone to time
-     out to silence, rather than continuing forever.
-
- ### res_pjsip: Enable TLS v1.3 if present.
-     res_pjsip now allows TLS v1.3 to be enabled if supported by the underlying
-     PJSIP library. The bundled version of PJSIP supports TLS v1.3.
-
- ### app_queue: Add support for applying caller priority change immediately.
-     The 'queue priority caller' CLI command and 'QueueChangePriorityCaller'
-     AMI action now have an 'immediate' argument which allows the caller priority
-     change to be reflected immediately, causing the position of a caller to move
-     within the queue depending on the priorities of the other callers.
-
- ### Adds manager actions to allow move/remove/forward individual messages in a
-     particular mailbox folder. The forward command can be used to copy a message
-     within a mailbox or to another mailbox. Also adds a VoicemailBoxSummarry,
-     required to retrieve message ID's.
-     The following manager actions have been added
-     VoicemailBoxSummary - Generate message list for a given mailbox
-     VoicemailRemove - Remove a message from a mailbox folder
-     VoicemailMove - Move a message from one folder to another within a mailbox
-     VoicemailForward - Copy a message from one folder in one mailbox to
-     another folder in another or the same mailbox.
-
- ### app_voicemail: add CLI commands for message manipulation
-     The following CLI commands have been added to app_voicemail
-     voicemail show mailbox &lt;mailbox&gt; &lt;context&gt;
-     Show contents of mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail remove &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt;
-     Remove message &lt;messageid&gt; from &lt;from_folder&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt;
-     voicemail move &lt;mailbox&gt; &lt;context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_folder&gt;
-     Move message &lt;messageid&gt; in mailbox &lt;mailbox&gt;&amp;&lt;context&gt; from &lt;from_folder&gt; to &lt;to_folder&gt;
-     voicemail forward &lt;from_mailbox&gt; &lt;from_context&gt; &lt;from_folder&gt; &lt;messageid&gt; &lt;to_mailbox&gt; &lt;to_context&gt; &lt;to_folder&gt;
-     Forward message &lt;messageid&gt; in mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;from_folder&gt; to mailbox &lt;mailbox&gt;@&lt;context&gt; &lt;to_folder&gt;
-
- ### sig_analog: Allow immediate fake ring to be suppressed.
-     The immediatering option can now be set to no to suppress the fake audible
-     ringback provided when immediate=yes on FXS channels.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #71: [new-feature]: core/ari/pjsip: Add refer mechanism to refer endpoints to
-      some resource
- #118: [new-feature]: chan_dahdi: Allow fake ringing to be inhibited when
-       immediate=yes
- #170: [improvement]: app_voicemail - add CLI commands to manipulate messages
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #181: [improvement]: app_voicemail - add manager actions to display and
-       manipulate messages
- #202: [improvement]: app_queue: Add support for immediately applying queue
-       caller priority change
- #205: [new-feature]: sig_analog: Allow flash to time out to silent hold
- #224: [new-feature]: chan_dahdi: Allow automatic reorigination on hangup
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #233: [bug]: Deadlock with MixMonitorMute AMI action
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #265: [bug]: app_macro isn't locking around channel datastore access
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630401">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
</package>
<package pkgid="02db7c69d6a5eb2ccbc57d610f5e98c84fd49daad09d7d2d7880e2c988af5859" name="asterisk" arch="src">
  <version epoch="0" ver="21.4.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.0-1" date="1697630400">- The Asterisk Development Team would like to announce
- the release of asterisk-21.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.0
- and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community
- and would have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.0
- ======================================
-
- Links:
- ------
-
- [Full
- ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.0.md)
- [GitHub
- Diff](https://github.com/asterisk/asterisk/compare/21.0.0-pre1...21.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Update master branch for Asterisk 21
- translate.c: Prefer better codecs upon translate ties.
- chan_skinny: Remove deprecated module.
- app_osplookup: Remove deprecated module.
- chan_mgcp: Remove deprecated module.
- chan_alsa: Remove deprecated module.
- pbx_builtins: Remove deprecated and defunct functionality.
- chan_sip: Remove deprecated module.
- app_cdr: Remove deprecated application and option.
- app_macro: Remove deprecated module.
- res_monitor: Remove deprecated module.
- http.c: Minor simplification to HTTP status output.
- app_osplookup: Remove obsolete sample config.
- say.c: Fix French time playback. (#42)
- core: Cleanup gerrit and JIRA references. (#58)
- utils.h: Deprecate `ast_gethostbyname()`. (#79)
- res_pjsip_pubsub: Add new pubsub module capabilities. (#82)
- app_sla: Migrate SLA applications out of app_meetme.
- rest-api: Ran make ari stubs to fix resource_endpoints inconsistency
- .github: Update AsteriskReleaser for security releases
- users.conf: Deprecate users.conf configuration.
- Update version for Asterisk 21
- Remove unneeded CHANGES and UPGRADE files
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix more local anchor references
- ari-stubs: Fix broken documentation anchors
- res_pjsip_session: Send Session Interval too small response
- .github: Update workflow-application-token-action to v2
- app_dial: Fix infinite loop when sending digits.
- app_voicemail: Fix for loop declarations
- alembic: Fix quoting of the 100rel column
- pbx.c: Fix gcc 12 compiler warning.
- app_audiosocket: Fixed timeout with -1 to avoid busy loop.
- download_externals: Fix a few version related issues
- main/refer.c: Fix double free in refer_data_destructor + potential leak
- sig_analog: Add Called Subscriber Held capability.
- Revert "app_stack: Print proper exit location for PBXless channels."
- install_prereq: Fix dependency install on aarch64.
- res_pjsip.c: Set contact_user on incoming call local Contact header
- extconfig: Allow explicit DB result set ordering to be disabled.
- rest-api: Run make ari-stubs
- res_pjsip_header_funcs: Make prefix argument optional.
- pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
- manager: Tolerate stasis messages with no channel snapshot.
- Remove unneeded CHANGES and UPGRADE files
-
- User Notes:
- -----------
-
- ### sig_analog: Add Called Subscriber Held capability.
-     Called Subscriber Held is now supported for analog FXS channels, using the
-     calledsubscriberheld option. This allows a station  user to go on hook when
-     receiving an incoming call and resume from another phone on the same line
-     by going on hook, without disconnecting the call.
-
- ### res_pjsip_header_funcs: Make prefix argument optional.
-     The prefix argument to PJSIP_HEADERS is now optional. If not specified,
-     all header names will be returned.
-
- ### http.c: Minor simplification to HTTP status output.
-     For bound addresses, the HTTP status page now combines the bound address
-     and bound port in a single line. Additionally, the SSL bind address has
-     been renamed to TLS.
-
- Upgrade Notes:
- --------------
-
- ### utils.h: Deprecate `ast_gethostbyname()`. (#79)
-     ast_gethostbyname() has been deprecated and will be removed in Asterisk
-     23. New code should use `ast_sockaddr_resolve()` and
-     `ast_sockaddr_resolve_first_af()`.
-
- ### app_sla: Migrate SLA applications out of app_meetme.
-     The SLAStation and SLATrunk applications have been moved from app_meetme
-     to app_sla. If you are using these applications and have autoload=no, you
-     will need to explicitly load this module in modules.conf.
-
- ### users.conf: Deprecate users.conf configuration.
-     The users.conf config is now deprecated and will be removed in a future
-     version of Asterisk.
-
- ### res_monitor: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. This also removes
-     the 'w' and 'W' options for app_queue. MixMonitor should be default and
-     only option for all settings that previously used either Monitor or
-     MixMonitor.
-
- ### app_osplookup: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### app_cdr: Remove deprecated application and option.
-     The previously deprecated NoCDR application has been removed.
-     Additionally, the previously deprecated 'e' option to the ResetCDR
-     application has been removed.
-
- ### app_macro: Remove deprecated module.
-     This module was deprecated in Asterisk 16 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy. For most modules
-     that interacted with app_macro, this change is limited to no longer looking
-     for the current context from the macrocontext when set. The following
-     modules have additional impacts:
-     app_dial - no longer supports M^ connected/redirecting macro
-     app_minivm - samples written using macro will no longer work.
-     The sample needs to be re-written
-     app_queue - can no longer call a macro on the called party's channel. Use
-     gosub which is currently supported
-     ccss - no callback macro, gosub only
-     app_voicemail - no macro support
-     channel - remove macrocontext and priority, no connected line or
-     redirection macro options
-     options - stdexten is deprecated to gosub as the default and only options
-     pbx - removed macrolock
-     pbx_dundi - no longer look for macro
-     snmp - removed macro context, exten, and priority
-
- ### translate.c: Prefer better codecs upon translate ties.
-     When setting up translation between two codecs the quality was not taken
-     into account, resulting in suboptimal translation. The quality is now taken
-     into account, which can reduce the number of translation steps required,
-     and improve the resulting quality.
-
- ### chan_sip: Remove deprecated module.
-     This module was deprecated in Asterisk 17 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_alsa: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### pbx_builtins: Remove deprecated and defunct functionality.
-     The previously deprecated ImportVar and SetAMAFlags applications have now
-     been removed.
-
- ### chan_mgcp: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- ### chan_skinny: Remove deprecated module.
-     This module was deprecated in Asterisk 19 and is now being removed in
-     accordance with the Asterisk Module Deprecation policy.
-
- Closed Issues:
- --------------
-
- #37: [Bug]: contrib/scripts/install_prereq tries to install armhf packages on
-      aarch64 Debian platforms
- #39: [Bug]: Remove .gitreview from repository.
- #41: [Bug]: say.c Time announcement does not say o'clock for the French
-      language
- #50: [improvement]: app_sla: Migrate SLA applications from app_meetme
- #78: [improvement]: Deprecate ast_gethostbyname()
- #81: [improvement]: Enhance and add additional PJSIP pubsub callbacks
- #179: [bug]: Queue strategy “Linear” with Asterisk 20 on Realtime
- #183: [deprecation]: Deprecate users.conf
- #226: [improvement]: Apply contact_user to incoming calls
- #230: [bug]: PJSIP_RESPONSE_HEADERS function documentation is misleading
- #240: [new-feature]: sig_analog: Add Called Subscriber Held capability
- #253: app_gosub patch appear to have broken predial handlers that utilize
-       macros that call gosubs
- #255: [bug]: pjsip_endpt_register_module: Assertion "Too many modules
-       registered"
- #263: [bug]: download_externals doesn't always handle versions correctly
- #267: [bug]: ari: refer with display_name key in request body leads to crash
- #274: [bug]: Syntax Error in SQL Code
- #275: [bug]:Asterisk make now requires ASTCFLAGS='-std=gnu99
-       -Wdeclaration-after-statement'
- #277: [bug]: pbx.c: Compiler error with gcc 12.2
- #281: [bug]: app_dial: Infinite loop if called channel hangs up while
-       receiving digits
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-       build_resource_tree</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
</package>
<package pkgid="d3c3763dd501d95ca12837f7aec271a6db29d8150ed12ba038dadf3cae68dc83" name="asterisk" arch="src">
  <version epoch="0" ver="21.4.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.1-1" date="1702555200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.0.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- The following security advisories were resolved in this release:
- - [Path traversal via AMI GetConfig allows access to outside files]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-8857-hfmw-vg8f)
- - [Asterisk susceptible to Denial of Service via DTLS Hello packets during call
-   initiation](https://github.com/asterisk/asterisk/security/advisories/GHSA-hxj9-xwr8-w8pq)
- - [PJSIP logging allows attacker to inject fake Asterisk log entries]
-   (https://github.com/asterisk/asterisk/security/advisories/GHSA-5743-x3p5-3rg7)
- - [PJSIP_HEADER dialplan function can overwrite memory/cause crash when using
-   'update'](https://github.com/asterisk/asterisk/security/advisories/GHSA-98rc-4j27-74hh)
-
- Change Log for Release asterisk-21.0.1
- ======================================
-
- Links:
- ------
-
- - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.1.md)
- - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.0...21.0.1)
- - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.1.tar.gz)
- - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_pjsip_header_funcs: Duplicate new header value, don't copy.
- - res_pjsip: disable raw bad packet logging
- - res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
- - manager.c: Prevent path traversal with GetConfig.
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
</package>
<package pkgid="5f406e8d54cbc0ac1787087afbbf3e30ccddcf86435a0d7e41d252a0aa9b900a" name="asterisk" arch="src">
  <version epoch="0" ver="21.5.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.0.2-1" date="1703073600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.0.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.0.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.0.2
- =======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.0.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.1...21.0.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.0.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- - res_rtp_asterisk: Fix regression issues with DTLS client check
-
- User Notes:
- -----------
-
- Upgrade Notes:
- --------------
-
- Closed Issues:
- --------------
-
- #500: [bug regression]: res_rtp_asterisk doesn't build if pjproject isn't used
- #503: [bug]: The res_rtp_asterisk DTLS check against ICE candidates fails when
-       it shouldn't
- #505: [bug]: res_pjproject: ast_sockaddr_cmp() always fails on sockaddrs
-       created by ast_sockaddr_from_pj_sockaddr()</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
</package>
<package pkgid="dbfc22d10b9b5c9c39b826e9504c1c438cc88f1417597070f893c72f58b81257" name="asterisk" arch="src">
  <version epoch="0" ver="21.6.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="b07ad67aa5d0be5827bddb90501f02435e6d4a6f7c4cd671d17ff0fbcd61ad75" name="asterisk" arch="src">
  <version epoch="0" ver="21.6.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="c8d0d37d68e85ce699f1b9f014ca2eeed3a7d7b9ee50f1019e486fa7641ca815" name="asterisk" arch="src">
  <version epoch="0" ver="21.7.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="ec6c634ebd2e9f7f2343ae375c5733ff647bf769900aedd35c8fef7430a61116" name="asterisk" arch="src">
  <version epoch="0" ver="21.8.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="8aa591bd22444437d288ecd5ea3093c9e33caa1dea582b26f5de7563834c3f5d" name="asterisk" arch="src">
  <version epoch="0" ver="21.9.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.6.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.6.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.6.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk 21.6.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.6.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.6.1
-
- Change Log for Release asterisk-21.6.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.0...21.6.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- - !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                         outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.7.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.7.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.7.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.7.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.7.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.7.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.6.1...21.7.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.7.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.8.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.8.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.8.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.8.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.8.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.8.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.7.0...21.8.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.8.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- -------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                  PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.9.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-21.9.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.9.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.9.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.9.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.9.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.8.0...21.9.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.9.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="ff2888587061c8ea9c13e89cc84009390760ea32cb09ba739c223995492817d5" name="asterisk" arch="src">
  <version epoch="0" ver="22.0.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.1.0-1" date="1706184000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.0.2...21.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- logger: Fix linking regression.
- Revert "core &amp; res_pjsip: Improve topology change handling."
- menuselect: Use more specific error message.
- res_pjsip_nat: Fix potential use of uninitialized transport details
- app_if: Fix faulty EndIf branching.
- manager.c: Fix regression due to using wrong free function.
- doc: Remove obsolete CHANGES-staging and UPGRADE-staging directories
- config_options.c: Fix truncation of option descriptions.
- manager.c: Improve clarity of "manager show connected".
- make_xml_documentation: Really collect LOCAL_MOD_SUBDIRS documentation.
- general: Fix broken links.
- MergeApproved.yml: Remove unneeded concurrency
- app_dial: Add option "j" to preserve initial stream topology of caller
- pbx_config.c: Don't crash when unloading module.
- ast_coredumper: Increase reliability
- logger.c: Move LOG_GROUP documentation to dedicated XML file.
- res_odbc.c: Allow concurrent access to request odbc connections
- res_pjsip_header_funcs.c: Check URI parameter length before copying.
- config.c: Log #exec include failures.
- make_xml_documentation: Properly handle absolute LOCAL_MOD_SUBDIRS.
- app_voicemail.c: Completely resequence mailbox folders.
- sig_analog: Fix channel leak when mwimonitor is enabled.
- res_rtp_asterisk.c: Update for OpenSSL 3+.
- alembic: Update list of TLS methods available on ps_transports.
- func_channel: Expose previously unsettable options.
- app.c: Allow ampersands in playback lists to be escaped.
- uri.c: Simplify ast_uri_make_host_with_port()
- func_curl.c: Remove CURLOPT() plaintext documentation.
- res_http_websocket.c: Set hostname on client for certificate validation.
- live_ast: Add astcachedir to generated asterisk.conf.
- SECURITY.md: Update with correct documentation URL
- func_lock: Add missing see-also refs to documentation.
- app_followme.c: Grab reference on nativeformats before using it
- configs: Improve documentation for bandwidth in iax.conf.
- logger: Add channel-based filtering.
- chan_iax2.c: Don't send unsanitized data to the logger.
- codec_ilbc: Disable system ilbc if version &gt;= 3.0.0
- resource_channels.c: Explicit codec request when creating UnicastRTP.
- doc: Update IP Quality of Service links.
- chan_pjsip: Add PJSIPHangup dialplan app and manager action
- chan_iax2.c: Ensure all IEs are displayed when dumping frame contents.
- chan_dahdi: Warn if nonexistent cadence is requested.
- stasis: Update the snapshot after setting the redirect
- ari: Provide the caller ID RDNIS for the channels
- main/utils: Implement ast_get_tid() for OpenBSD
- res_rtp_asterisk.c: Fix runtime issue with LibreSSL
- app_directory: Add ADSI support to Directory.
- core_local: Fix local channel parsing with slashes.
- Remove files that are no longer updated
- app_voicemail: Add AMI event for mailbox PIN changes.
- app_queue.c: Emit unpause reason with PauseQueueMember event.
- bridge_simple: Suppress unchanged topology change requests
- res_pjsip: Include cipher limit in config error message.
- res_speech: allow speech to translate input channel
- res_rtp_asterisk.c: Fix memory leak in ephemeral certificate creation.
- res_pjsip_dtmf_info.c: Add 'INFO' to Allow header.
- api.wiki.mustache: Fix indentation in generated markdown
- pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
- configs: Fix typo in pjsip.conf.sample.
- res_pjsip_exten_state,res_pjsip_mwi: Allow unload on shutdown
- res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-            characters
- .github: PRSubmitActions: Fix adding reviewers to PR
- .github: New PR Submit workflows
- .github: New PR Submit workflows
- res_stasis: signal when new command is queued
- ari/stasis: Indicate progress before playback on a bridge
- func_curl.c: Ensure channel is locked when manipulating datastores.
- .github: Fix job prereqs in PROpenedUpdated
- .github: Block PR tests until approved
- .github: Use generic releaser
- logger.h: Add ability to change the prefix on SCOPE_TRACE output
- Add libjwt to third-party
- res_pjsip: update qualify_timeout documentation with DNS note
- chan_dahdi: Clarify scope of callgroup/pickupgroup.
- func_json: Fix crashes for some types
- res_speech_aeap: add aeap error handling
- app_voicemail: Disable ADSI if unavailable.
- codec_builtin: Use multiples of 20 for maximum_ms
- lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
- asterisk.c: Use the euid's home directory to read/write cli history
- res_pjsip_transport_websocket: Prevent transport from being destroyed before
-                                message finishes.
- cel: add publish user event helper
- chan_console: Fix deadlock caused by unclean thread exit.
- file.c: Add ability to search custom dir for sounds
- chan_iax2: Improve authentication debugging.
- res_rtp_asterisk: fix wrong counter management in ioqueue objects
- res_pjsip_pubsub: Add body_type to test_handler for unit tests
- make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
- func_periodic_hook: Add hangup step to avoid timeout
- res_stasis_recording.c: Save recording state when unmuted.
- res_speech_aeap: check for null format on response
- func_periodic_hook: Don't truncate channel name
- safe_asterisk: Change directory permissions to 755
- chan_rtp: Implement RTP glue for UnicastRTP channels
- app_queue: periodic announcement configurable start time.
- variables: Add additional variable dialplan functions.
- Restore CHANGES and UPGRADE.txt to allow cherry-picks to work
-
- User Notes:
- -----------
-
- ### app_dial: Add option "j" to preserve initial stream topology of caller
-               The option "j" is now available for the Dial application which
-               uses the initial stream topology of the caller to create the
-               outgoing channels.
-
- ### logger: Add channel-based filtering.
-             The console log can now be filtered by channels or groups of
-             channels, using the logger filter CLI commands.
-
- ### chan_pjsip: Add PJSIPHangup dialplan app and manager action
-                 A new dialplan app PJSIPHangup and AMI action allows you to
-                 hang up an unanswered incoming PJSIP call with a specific SIP
-                 response code in the 400 -&gt; 699 range.
-
- ### app_voicemail: Add AMI event for mailbox PIN changes.
-                    The VoicemailPasswordChange event is now emitted whenever a
-                    mailbox password is updated, containing the mailbox
-                    information and the new password. Resolves: #398
-
- ### res_speech: allow speech to translate input channel
-                 res_speech now supports translation of an input channel to a
-                 format supported by the speech provider, provided a translation
-                 path is available between the source format and provider
-                 capabilites.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                With this update, the PJSIP realm lengths have been extended to
-                support up to 255 characters.
-
- ### res_stasis: signal when new command is queued
-                 Call setup times should be significantly improved when using
-                 ARI.
-
- ### lock.c: Separate DETECT_DEADLOCKS from DEBUG_THREADS
-             You no longer need to select DEBUG_THREADS to use DETECT_DEADLOCKS.
-             This removes a significant amount of overhead if you just want to
-             detect possible deadlocks vs needing full lock tracing.
-
- ### file.c: Add ability to search custom dir for sounds
-             A new option "sounds_search_custom_dir" has been added to
-             asterisk.conf that allows asterisk to search
-             AST_DATA_DIR/sounds/custom for sounds files before searching the
-             standard AST_DATA_DIR/sounds/&lt;lang&gt; directory.
-
- ### make_buildopts_h, et. al. Allow adding all cflags to buildopts.h
-                               The "Build Options" entry in the "core show
-                               settings" CLI command has been renamed to "ABI
-                               related Build Options" and a new entry named "All
-                               Build Options" has been added that shows both
-                               breaking and non-breaking options.
-
- ### chan_rtp: Implement RTP glue for UnicastRTP channels
-               The dial string option 'g' was added to the UnicastRTP channel
-               which enables RTP glue and therefore native RTP bridges with
-               those channels.
-
- ### app_queue: periodic announcement configurable start time.
-                Introduce a new queue configuration option called
-                'periodic-announce-startdelay' which will vary the normal
-                (historic) behavior of starting the periodic announcement cycle
-                at periodic-announce-frequency seconds after entering the queue
-                to start the periodic announcement cycle at
-                period-announce-startdelay seconds after joining the queue. The
-                default behavior if this config option is not set remains
-                unchanged. Signed-off-by: Jaco Kroon &lt;jaco@uls.co.za&gt;
-
- ### variables: Add additional variable dialplan functions.
-                Four new dialplan functions have been added. GLOBAL_DELETE and
-                DELETE have been added which allows the deletion of global and
-                channel variables. GLOBAL_EXISTS and VARIABLE_EXISTS have been
-                added which checks whether a global or channel variable has been
-                set.
-
- Upgrade Notes:
- --------------
-
- ### app.c: Allow ampersands in playback lists to be escaped.
-            Ampersands in URLs passed to the `Playback()`, `Background()`,
-            `SpeechBackground()`, `Read()`, `Authenticate()`, or `Queue()`
-            applications as filename arguments can now be escaped by single
-            quoting the filename. Additionally, this is also possible when using
-            the `CONFBRIDGE` dialplan function, or configuring various features
-            in `confbridge.conf` and `queues.conf`.
-
- ### pjsip_configuration.c: Disable DTLS renegotiation if WebRTC is enabled.
-                            The dtls_rekey will be disabled if webrtc support is
-                            requested on an endpoint. A warning will also be
-                            emitted.
-
- ### res_pjsip: Expanding PJSIP endpoint ID and relevant resource length to 255
-                characters
-                As part of this update, the maximum allowable length for PJSIP
-                endpoints and relevant resources has been increased from 40 to
-                255 characters. To take advantage of this enhancement, it is
-                recommended to run the necessary procedures (e.g., Alembic) to
-                update your schemas.
-
- Closed Issues:
- --------------
-
- #84: [bug]: codec_ilbc: Fails to build with ilbc version 3.0.4
- #129: [bug]: res_speech_aeap: Crash due to NULL format on setup
- #242: [new-feature]: logger: Allow filtering logs in CLI by channel
- #248: [bug]: core_local: Local channels cannot have slashes in the destination
- #260: [bug]: maxptime must be changed to multiples of 20
- #286: [improvement]: chan_iax2: Improve authentication debugging
- #289: [new-feature]: Add support for deleting channel and global variables
- #294: [improvement]: chan_dahdi: Improve call pickup documentation
- #298: [improvement]: chan_rtp: Implement RTP glue
- #301: [bug]: Number of ICE TURN threads continually growing
- #303: [bug]: SpeechBackground never exits
- #308: [bug]: chan_console: Deadlock when hanging up console channels
- #315: [improvement]: Search /var/lib/asterisk/sounds/custom for sound files
-                      before /var/lib/asterisk/sounds/&lt;lang&gt;
- #316: [bug]: Privilege Escalation in Astrisk's Group permissions.
- #319: [bug]: func_periodic_hook truncates long channel names when setting
-              EncodedChannel
- #321: [bug]: Performance suffers unnecessarily when debugging deadlocks
- #325: [bug]: hangup after beep to avoid waiting for timeout
- #330: [improvement]: Add cel user event helper function
- #335: [bug]: res_pjsip_pubsub: The bad_event unit test causes a SEGV in
-              build_resource_tree
- #337: [bug]: asterisk.c: The CLI history file is written to the wrong directory
-              in some cases
- #341: [bug]: app_if.c : nested EndIf incorrectly exits parent If
- #345: [improvement]: Increase pj_sip Realm Size to 255 Characters for Improved
-                      Functionality
- #349: [improvement]: Add libjwt to third-party
- #352: [bug]: Update qualify_timeout documentation to include DNS note
- #354: [improvement]: app_voicemail: Disable ADSI if unavailable on a line
- #356: [new-feature]: app_directory: Add ADSI support.
- #360: [improvement]: Update documentation for CHANGES/UPGRADE files
- #362: [improvement]: Speed up ARI command processing
- #379: [bug]: Orphaned taskprocessors cause shutdown delays
- #384: [bug]: Unnecessary re-INVITE after answer
- #388: [bug]: Crash in app_followme.c due to not acquiring a reference to
-              nativeformats
- #396: [improvement]: res_pjsip: Specify max ciphers allowed if too many
-                      provided
- #398: [new-feature]: app_voicemail: Add AMI event for password change
- #409: [improvement]: chan_dahdi: Emit warning if specifying nonexistent cadence
- #423: [improvement]: func_lock: Add missing see-also refs
- #425: [improvement]: configs: Improve documentation for bandwidth in
-                      iax.conf.sample
- #428: [bug]: cli: Output is truncated from "config show help"
- #430: [bug]: Fix broken links
- #442: [bug]: func_channel: Some channel options are not settable
- #445: [bug]: ast_coredumper isn't figuring out file locations properly in all
-              cases
- #458: [bug]: Memory leak in chan_dahdi when mwimonitor=yes on FXO
- #462: [new-feature]: app_dial: Add new option to preserve initial stream
-                      topology of caller
- #465: [improvement]: Change res_odbc connection pool request logic to not lock
-                      around blocking operations
- #482: [improvement]: manager.c: Improve clarity of "manager show connected"
-                      output
- #509: [bug]: res_pjsip: Crash when looking up transport state in use
- #513: [bug]: manager.c: Crash due to regression using wrong free function when
-                         built with MALLOC_DEBUG
- #520: [improvement]: menuselect: Use more specific error message.
- #530: [bug]: bridge_channel.c: Stream topology change amplification with
-              multiple layers of Local channels
- #539: [bug]: Existence of logger.xml causes linking failure</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
</package>
<package pkgid="a396d4334e2cfd088d4d0d39652377422e0ed2d717193a1d82e8ce926869c639" name="asterisk" arch="src">
  <version epoch="0" ver="22.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.2.0-1" date="1710763200">- The Asterisk Development Team would like to announce the release of
- asterisk-21.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.2.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.1.0...21.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- res_pjsip_stir_shaken.c:  Add checks for missing parameters
- app_dial: Add dial time for progress/ringing.
- app_voicemail: Properly reinitialize config after unit tests.
- app_queue.c : fix "queue add member" usage string
- app_voicemail: Allow preventing mark messages as urgent.
- res_pjsip: Use consistent type for boolean columns.
- attestation_config.c: Use ast_free instead of ast_std_free
- Makefile: Add stir_shaken/cache to directories created on install
- Stir/Shaken Refactor
- translate.c: implement new direct comp table mode
- README.md: Removed outdated link
- strings.h: Ensure ast_str_buffer(…) returns a 0 terminated string.
- res_rtp_asterisk.c: Correct coefficient in MOS calculation.
- dsp.c: Fix and improve potentially inaccurate log message.
- pjsip show channelstats: Prevent possible segfault when faxing
- Reduce startup/shutdown verbose logging
- configure: Rerun bootstrap on modern platform.
- Upgrade bundled pjproject to 2.14.
- res_pjsip_outbound_registration.c: Add User-Agent header override
- app_speech_utils.c: Allow partial speech results.
- utils: Make behavior of ast_strsep* match strsep.
- app_chanspy: Add 'D' option for dual-channel audio
- app_if: Fix next priority calculation.
- res_pjsip_t38.c: Permit IPv6 SDP connection addresses.
- BuildSystem: Bump autotools versions on OpenBSD.
- main/utils: Simplify the FreeBSD ast_get_tid() handling
- res_pjsip_session.c: Correctly format SDP connection addresses.
- rtp_engine.c: Correct sample rate typo for L16/44100.
- manager.c: Fix erroneous reloads in UpdateConfig.
- res_calendar_icalendar: Print iCalendar error on parsing failure.
- app_confbridge: Don't emit warnings on valid configurations.
- app_voicemail_odbc: remove macrocontext from voicemail_messages table
- chan_dahdi: Allow MWI to be manually toggled on channels.
- chan_rtp.c: MulticastRTP missing refcount without codec option
- chan_rtp.c: Change MulticastRTP nameing to avoid memory leak
- func_frame_trace: Add CLI command to dump frame queue.
-
- User Notes:
- -----------

- ### app_dial: Add dial time for progress/ringing.
-               The timeout argument to Dial now allows specifying the maximum
-               amount of time to dial if early media is not received.
-
- ### app_voicemail: Allow preventing mark messages as urgent.
-                    The leaveurgent mailbox option can now be used to control
-                    whether callers may leave messages marked as 'Urgent'.
-
- ### Stir/Shaken Refactor
-     Asterisk's stir-shaken feature has been refactored to correct
-     interoperability, RFC compliance, and performance issues. See
-     https://docs.asterisk.org/Deployment/STIR-SHAKEN for more information.
-
- ### Upgrade bundled pjproject to 2.14.
-     Bundled pjproject has been upgraded to 2.14. For more information on what
-     all is included in this change, check out the pjproject Github page:
-     https://github.com/pjsip/pjproject/releases
-
- ### res_pjsip_outbound_registration.c: Add User-Agent header override
-                                        PJSIP outbound registrations now
-                                        support a per-registration User-Agent
-                                        header
-
- ### app_speech_utils.c: Allow partial speech results.
-                         The SpeechBackground dialplan application now supports
-                         a 'p' option that will return partial results from
-                         speech engines that provide them when a timeout
-                         occurs.
-
- ### app_chanspy: Add 'D' option for dual-channel audio
-                  The ChanSpy application now accepts the 'D' option which will
-                  interleave the spied audio within the outgoing frames. The
-                  purpose of this is to allow the audio to be read as a Dual
-                  channel stream with separate incoming and outgoing audio.
-                  Setting both the 'o' option and the 'D' option and results in
-                  the 'D' option being ignored.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-                         The fix requires removing the macrocontext column from
-                         the voicemail_messages table in the voicemail database
-                         via alembic upgrade.
-
- ### chan_dahdi: Allow MWI to be manually toggled on channels.
-                 The 'dahdi set mwi' now allows MWI on channels to be manually
-                 toggled if needed for troubleshooting. Resolves: #440
-
- Upgrade Notes:
- --------------
-
- ### Stir/Shaken Refactor
-     The stir-shaken refactor is a breaking change but since it's not working
-     now we don't think it matters. The stir_shaken.conf file has changed
-     significantly which means that existing ones WILL need to be changed. The
-     stir_shaken.conf.sample file in configs/samples/ has quite a bit more
-     information. This is also an ABI breaking change since some of the
-     existing objects needed to be changed or removed, and new ones added.
-     Additionally, if res_stir_shaken is enabled in menuselect, you'll need to
-     either have the development package for libjwt v1.15.3 installed or use
-     the --with-libjwt-bundled option with ./configure.
-
- ### app_voicemail_odbc: remove macrocontext from voicemail_messages table
-     The fix requires that the voicemail database be upgraded via alembic.
-     Upgrading to the latest voicemail database via alembic will remove the
-     macrocontext column from the voicemail_messages table.
-
- Closed Issues:
- --------------
-
- #46: [bug]: Stir/Shaken: Wrong CID used when looking up certificates
- #351: [improvement]: Refactor res_stir_shaken to use libjwt
- #406: [improvement]: pjsip: Upgrade bundled version to pjproject 2.14
- #440: [new-feature]: chan_dahdi: Allow manually toggling MWI on channels
- #492: [improvement]: res_calendar_icalendar: Print icalendar error if
-                      available on parsing failure
- #515: [improvement]: Implement option to override User-Agent-Header on a
-                      per-registration basis
- #527: [bug]: app_voicemail_odbc no longer working after removal of
-              macrocontext.
- #529: [bug]: MulticastRTP without selected codec leeds to "FRACK!, Failed
-              assertion bad magic number 0x0 for object" after ~30 calls
- #533: [improvement]: channel.c, func_frame_trace.c: Improve debuggability of
-                      channel frame queue
- #551: [bug]: manager: UpdateConfig triggers reload with "Reload: no"
- #560: [bug]: EndIf() causes next priority to be skipped
- #565: [bug]: Application Read() returns immediately
- #569: [improvement]: Add option to interleave input and output frames on
-                      spied channel
- #572: [improvement]: Copy partial speech results when Asterisk is ready to
-                      move on but the speech backend is not
- #582: [improvement]: Reduce unneeded logging during startup and shutdown
- #586: [bug]: The "restrict" keyword used in chan_iax2.c isn't supported in
-              older gcc versions
- #588: [new-feature]: app_dial: Allow Dial to be aborted if early media is not
-                      received
- #592: [bug]: In certain circumstances, "pjsip show channelstats" can segfault
-              when a fax session is active
- #595: [improvement]: dsp.c: Fix and improve confusing warning message.
- #597: [bug]: wrong MOS calculation
- #601: [new-feature]: translate.c: implement new direct comp table mode (PR
-                      #585)
- #619: [new-feature]: app_voicemail: Allow preventing callers from marking
-                      messages as urgent
- #629: [bug]: app_voicemail: Multiple executions of unit tests cause segfault
- #634: [bug]: make install doesn't create the stir_shaken cache directory
- #636: [bug]: Possible SEGV in res_stir_shaken due to wrong free function
- #645: [bug]: Occasional SEGV in res_pjsip_stir_shaken.c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
</package>
<package pkgid="51be95d3b2e595821fd0bd91689ae9e54f639980036df4dd57a00e2946256a10" name="asterisk" arch="src">
  <version epoch="0" ver="22.1.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.0-1" date="1715860800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.2.0...21.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 43
- Commit Authors: 15
- Issues Resolved: 26
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_logger: Preserve logging state on reloads.
-                        Issuing "pjsip reload" will no longer disable logging
-                        if it was previously enabled from the CLI.
-
- #### loader.c: Allow dependent modules to be unloaded recursively.
-                In certain circumstances, modules with dependency relations can
-                have their dependents automatically recursively unloaded and
-                loaded again using the "module refresh" CLI command or the
-                ModuleLoad AMI command.
-
- #### tcptls/iostream: Add support for setting SNI on client TLS connections
-                       Secure websocket client connections now send SNI in the
-                       TLS client hello.
-
- #### res_pjsip_endpoint_identifier_ip: Add endpoint identifier transport address.
-                                        set identify_by=transport for the pjsip
-                                        endpoint. Then use the existing 'match'
-                                        option and the new 'transport' option
-                                        of the identify. Fixes: #672
-
- #### res_pjsip_endpoint_identifier_ip: Endpoint identifier request URI
-                                        this new feature let users match
-                                        endpoints based on the indound SIP
-                                        requests' URI. To do so, add
-                                        'request_uri' to the endpoint's
-                                        'identify_by' option. The
-                                        'match_request_uri' option of the
-                                        identify can be an exact match for the
-                                        entire request uri, or a regular
-                                        expression (between slashes). It's
-                                        quite similar to the header identifer.
-                                        Fixes: #599
-
- #### res_pjsip_refer.c: Allow GET_TRANSFERRER_DATA
-                         the GET_TRANSFERRER_DATA dialplan variable can now be
-                         used also in pjsip.
-
- #### manager.c: Add new parameter 'PreDialGoSub' to Originate AMI action
-                 When using the Originate AMI Action, we now can pass the
-                 PreDialGoSub parameter, instructing the asterisk to perform an
-                 subrouting at channel before call start. With this parameter
-                 an call initiated by AMI can request the channel to start the
-                 call automaticaly, adding a SIP header to using GoSUB,
-                 instructing to autoanswer the channel, and proceeding the
-                 outbuound extension executing. Exemple of an context to
-                 perform the previus indication:
-                 [addautoanswer]
-                 exten =&gt; _s,1,Set(PJSIP_HEADER(add,Call-Info)=answer-after=0)
-                 exten =&gt; _s,n,Set(PJSIP_HEADER(add,Alert-Info)=answer-after=0)
-                 exten =&gt; _s,n,Return()
-
- #### manager.c: Add CLI command to kick AMI sessions.
-                 The "manager kick session" CLI command now allows kicking a
-                 specified AMI session.
-
- #### chan_dahdi: Allow specifying waitfordialtone per call.
-                  "waitfordialtone" may now be specified for DAHDI trunk
-                  channels on a per-call basis using the CHANNEL function.
-
- #### Upgrade bundled pjproject to 2.14.1
-      Bundled pjproject has been upgraded to 2.14.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.14.1
-
- Upgrade Notes:
- --------------
-
- #### pbx_variables.c: Prevent SEGV due to stack overflow.
-                       The maximum amount of dialplan recursion using variable
-                       substitution (such as by using EVAL_EXTEN) is capped at
-                       15.
-
- Commit Authors:
- ---------------
-
- Fabrice Fontaine: (1)
- George Joseph: (8)
- Henrik Liljedahl: (1)
- Holger Hans Peter Freyther: (1)
- Ivan Poddubny: (2)
- Jonatascalebe: (1)
- Joshua Elson: (1)
- Martin Nystroem: (1)
- Martin Tomec: (1)
- Maximilian Fridrich: (1)
- Naveen Albert: (14)
- Sean Bright: (7)
- Sperl Viktor: (2)
- Spiridonov Dmitry: (1)
- Stanislav Abramenkov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
</package>
<package pkgid="d041eb9c86d5f5cb022e134702b2e891cadcd2469c1a2a5df09ee28ef65f698f" name="asterisk" arch="src">
  <version epoch="0" ver="22.2.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.3.1-1" date="1715947200">- The Asterisk Development Team would like to announce security release Asterisk
- 21.3.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.3.1
- and https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.3.1
-
- Change Log for Release asterisk-21.3.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.3.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.0...21.3.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.3.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-qqxj-v78h-hrf9](https://github.com/asterisk/asterisk/security/advisories/GHSA-qqxj-v78h-hrf9):
-    res_pjsip_endpoint_identifier_ip: wrongly matches ALL unauthorized SIP
-    requests
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
</package>
<package pkgid="aba7826e91cb7f5c7bed3f19b59476958e000c1c6ed88e462d74549d6e17fbba" name="asterisk" arch="src">
  <version epoch="0" ver="22.3.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.0-1" date="1721304000">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.3.1...21.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 20
- Commit Authors: 9
- Issues Resolved: 8
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### app_voicemail_odbc: Allow audio to be kept on disk
-                          This commit adds a new voicemail.conf option
-                          'odbc_audio_on_disk' which when set causes the ODBC
-                          variant of app_voicemail_odbc to leave the message and
-                          greeting audio files on disk and only store the
-                          message metadata in the database. Much more
-                          information can be found in the voicemail.conf.sample
-                          file.
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a Queue option log-restricted-caller-id to control whether
-                 the Restricted Caller ID will be stored in the queue log. If
-                 log-restricted-caller-id=no then the Caller ID will be stripped
-                 if the Caller ID is restricted.
-
- #### pbx.c: expand fields width of "core show hints"
-             The fields width of "core show hints" were increased. The width of
-             "extension" field to 30 characters and the width of the "device
-             state id" field to 60 characters.
-
- #### rtp_engine: add support for multirate RFC2833 digits
-                  No change in configuration is required in order to enable this
-                  feature. Endpoints configured to use RFC2833 will
-                  automatically have this enabled. If the endpoint does not
-                  support this, it should not include it in the SDP
-                  offer/response.
-                  Resolves: #699
-
- Upgrade Notes:
- --------------
-
- #### app_queue: Add option to not log Restricted Caller ID to queue_log
-                 Add a new column to the queues table:
-                 queue_log_option_log_restricted
-                 ENUM('0','1','off','on','false','true','no','yes')
-                 to control whether the Restricted Caller ID will be stored in
-                 the queue log.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Bastian Triller: (1)
- Chrsmj: (1)
- George Joseph: (4)
- Igor Goncharovsky: (1)
- Mike Bradeen: (2)
- Sean Bright: (7)
- Tinet-Mucw: (1)
- Walter Doekes: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
</package>
<package pkgid="659cd3fd0bf512fc919fe10f3a76a7a1e3b7c871c98e6640e2a724c6a2f1380c" name="asterisk" arch="src">
  <version epoch="0" ver="22.4.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.1-1" date="1721908800">- The Asterisk Development Team would like to announce the release of
- asterisk-21.4.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.1
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.4.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.0...21.4.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 2
- Commit Authors: 1
- Issues Resolved: 2
- Security Advisories Resolved: 0
-
- Commit Authors:
- ---------------
-
- George Joseph: (2)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- 819: [bug]: Typo in voicemail.conf.sample that stops it from loading when using
-             "make samples"
- 822: [bug]: segfault in main/rtp_engine.c:1489 after updating 20.8.1 -&gt; 20.9.0
-
- Commits By Author:
- ------------------
-
- #### George Joseph (2):
-   voicemail.conf.sample: Fix ':' comment typo
-   rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-
- Commit List:
- ------------
-
- rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
- voicemail.conf.sample: Fix ':' comment typo
-
- Commit Details:
- ---------------
-
- #### rtp_engine.c: Prevent segfault in ast_rtp_codecs_payloads_unset()
-                    Author: George Joseph
-                    Date:   2024-07-25
-
-                    There can be empty slots in payload_mapping_tx corresponding
-                    to dynamic payload types that haven't been seen before so we
-                    now check for NULL before attempting to use 'type' in the
-                    call to ast_format_cmp.
-
-                    Note: Currently only chan_sip calls ast_rtp_codecs_payloads_unset()
-
-                    Resolves: #822
-
- #### voicemail.conf.sample: Fix ':' comment typo
-                             Author: George Joseph
-                             Date:   2024-07-24
-
-                             ...and removed an errant trailing space.
-
-                             Resolves: #819</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.2-1" date="1723118400">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.2.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.2 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.2
-
- Change Log for Release asterisk-21.4.2
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.2.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.1...21.4.2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
- - [GHSA-c4cg-9275-6w44](https://github.com/asterisk/asterisk/security/advisories/GHSA-c4cg-9275-6w44):
-                        Write=originate, is sufficient permissions for code
-                        execution / System() dialplan
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-c4cg-9275-6w44: Write=originate, is sufficient permissions for code
-                       execution / System() dialplan
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   manager.c: Add entries to Originate blacklist
-
- Commit List:
- ------------
-
- manager.c: Add entries to Originate blacklist
-
- Commit Details:
- ---------------
-
- #### manager.c: Add entries to Originate blacklist
-                 Author: George Joseph
-                 Date:   2024-07-22
-
-                 Added Reload and DBdeltree to the list of dialplan application
-                 that can't be executed via the Originate manager action without
-                 also having write SYSTEM permissions.
-
-                 Added CURL, DB*, FILE, ODBC and REALTIME* to the list of
-                 dialplan functions that can't be executed via the Originate
-                 manager action without also having write SYSTEM permissions.
-
-                 If the Queue application is attempted to be run by the
-                 Originate manager action and an AGI parameter is specified in
-                 the app data, it'll be rejected unless the manager user has
-                 either the AGI or SYSTEM permissions.
-
-                 Resolves: #GHSA-c4cg-9275-6w44</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.4.3-1" date="1725537600">- The Asterisk Development Team would like to announce security release Asterisk
- 21.4.3.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.4.3 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.4.3
-
- Change Log for Release asterisk-21.4.3
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.4.3.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.2...21.4.3)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.4.3.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-   [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                        A malformed Contact or Record-Route URI in an incoming
-                        SIP request can cause Asterisk to crash when
-                        res_resolver_unbound is used
-
- Commit Authors:
- ---------------
-
- George Joseph: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-v428-g3cw-7hv9: A malformed Contact or Record-Route URI in an incoming
-                       SIP request can cause Asterisk to crash when
-                       res_resolver_unbound is used
-
- Commits By Author:
- ------------------
-
- #### George Joseph (1):
-   res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit List:
- ------------
- res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-
- Commit Details:
- ---------------
-
- #### res_resolver_unbound: Test for NULL ub_result in unbound_resolver_callback
-                            Author: George Joseph
-                            Date:   2024-08-12
-
-                            The ub_result pointer passed to
-                            unbound_resolver_callback by libunbound can be NULL
-                            if the query was for something malformed like `.1`
-                            or `[.1]`. If it is, we now set a 'ns_r_formerr'
-                            result and return instead of crashing with a SEGV.
-                            This causes pjproject to simply cancel the
-                            transaction with a "No answer record in the DNS
-                            response" error. The existing "off nominal" unit
-                            test was also updated to check this condition.
-
-                            Although not necessary for this fix, we also made
-                            ast_dns_resolver_completed() tolerant of a NULL
-                            result.
-
-                            Resolves: GHSA-v428-g3cw-7hv9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 21.5.0-1" date="1729166400">- The Asterisk Development Team would like to announce the release of
- asterisk-21.5.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/21.5.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 21.5.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-21.5.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.5.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.4.3...21.5.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.5.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 8
- Issues Resolved: 17
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip_notify: add dialplan application
-                        A new dialplan application PJSIPNotify is now available
-                        which can send SIP NOTIFY requests from the dialplan.
-                        The pjsip send notify CLI command has also been enhanced
-                        to allow sending NOTIFY messages to a specific channel.
-                        Syntax: pjsip send notify &lt;option&gt; channel &lt;channel&gt;
-
- #### channel: Add multi-tenant identifier.
-               tenantid has been added to channels. It can be read in dialplan
-               via CHANNEL(tenantid), and it can be set using
-               Set(CHANNEL(tenantid)=My tenant ID). In pjsip.conf, it is
-               recommended to use the new tenantid option for pjsip endpoints
-               (e.g., tenantid=My tenant ID) so that it will show up in
-               Newchannel events. You can set it like any other channel variable
-               using set_var in pjsip.conf as well, but note that this will NOT
-               show up in Newchannel events. Tenant ID is also available in CDR
-               and can be accessed with CDR(tenantid). The peer tenant ID can
-               also be accessed with CDR(peertenantid). CEL includes tenant ID
-               as well if it has been set.
-
- #### res_pjsip_config_wizard.c: Refactor load process
-                                 The res_pjsip_config_wizard.so module can now
-                                 be reloaded.
-
- Upgrade Notes:
- --------------
-
- #### channel: Add multi-tenant identifier.
-               A new versioned struct (ast_channel_initializers) has been added
-               that gets passed to __ast_channel_alloc_ap. The new function
-               ast_channel_alloc_with_initializers should be used when creating
-               channels that require the use of this struct. Currently the only
-               value in the struct is for tenantid, but now more fields can be
-               added to the struct as necessary rather than the
-               __ast_channel_alloc_ap function. A new option (tenantid) has been
-               added to endpoints in pjsip.conf as well. CEL has had its version
-               bumped to include tenant ID.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- Ben Ford: (1)
- Cade Parker: (1)
- George Joseph: (11)
- Jaco Kroon: (1)
- Mike Bradeen: (3)
- Sean Bright: (2)
- Tinet-Mucw: (3)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.0.0-1" date="1729166401">- The Asterisk Development Team would like to announce the release of
- asterisk-22.0.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.0.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.0.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.0.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.0.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0-pre1...22.0.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.0.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 22
- Commit Authors: 8
- Issues Resolved: 13
- Security Advisories Resolved: 1
-  - [GHSA-v428-g3cw-7hv9](https://github.com/asterisk/asterisk/security/advisories/GHSA-v428-g3cw-7hv9):
-                         A malformed Contact or Record-Route URI in an incoming
-                         SIP request can cause Asterisk to crash when
-                         res_resolver_unbound is used
-
- User Notes:
- -----------
-
- #### feat: ARI "ChannelToneDetected" event
-            Setting the TONE_DETECT dialplan function on a channel in ARI will
-            now cause a ChannelToneDetected ARI event to be raised when the
-            specified tone is detected.
-
- Upgrade Notes:
- --------------
-
- #### app_record: Add RECORD_TIME output variable.
-                  The RECORD_TIME variable now contains the duration of Record()
-                  recordings in milliseconds.
-
- Commit Authors:
- ---------------
-
- Alexei Gradinari: (2)
- George Joseph: (10)
- Gibbz00: (1)
- Joshua C. Colp: (2)
- Mike Bradeen: (2)
- Naveen Albert: (1)
- Sean Bright: (2)
- Tinet-Mucw: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.0-1" date="1732190400">- The Asterisk Development Team would like to announce the release of
- asterisk-22.1.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.1.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.0.0...22.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-                 The new "suppress_moh_on_sendonly" endpoint option can be used
-                 to prevent playing MOH back to a caller if the remote end sends
-                 "sendonly" or "inactive" (hold) to Asterisk in an SDP.
-
- #### app_mixmonitor: Add 'D' option for dual-channel audio.
-                      The MixMonitor application now has a new 'D' option which
-                      interleaves the recorded audio in the output frames. This
-                      allows for stereo recording output with one channel being
-                      the transmitted audio and the other being the received
-                      audio. The 't' and 't' options are compatible with this.
-
- #### manager.c: Restrict ModuleLoad to the configured modules directory.
-                 The ModuleLoad AMI action now restricts modules to the
-                 configured modules directory.
-
- #### manager: Enhance event filtering for performance
-               You can now perform more granular filtering on events in
-               manager.conf using expressions like `eventfilter(name(Newchannel),header(Channel),method(starts_with))
-               = PJSIP/` This is much more efficient than `eventfilter = Event:
-               Newchannel.*Channel: PJSIP/` Full syntax guide is in
-               configs/samples/manager.conf.sample.
-
- #### db.c: Remove limit on family/key length
-            The `ast_db_*()` APIs have had the 253 byte limit on "/family/key"
-            removed and will now accept families and keys with a total length of
-            up to SQLITE_MAX_LENGTH (currently 1e9!). This affects the `DB*`
-            dialplan applications, dialplan functions, manager actions and
-            `databse` CLI commands. Since the media_cache also uses the
-            `ast_db_*()` APIs, you can now store resources with URIs longer than
-            253 bytes.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- Thomas Guebels: (2)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.1.1-1" date="1736424000">- The Asterisk Development Team would like to announce security release Asterisk
- 22.1.1.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.1.1 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.1.1
-
- Change Log for Release asterisk-22.1.1
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.1.1.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.0...22.1.1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.1.1.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 0
- Security Advisories Resolved: 1
-  - [GHSA-33x6-fj46-6rfh](https://github.com/asterisk/asterisk/security/advisories/GHSA-33x6-fj46-6rfh):
-    Path traversal via AMI ListCategories allows access to outside files
-
- User Notes:
- -----------
-
- #### manager.c: Restrict ListCategories to the configuration directory
-                 The ListCategories AMI action now restricts files to the
-                 configured configuration directory.
-
- Commit Authors:
- ---------------
-
- Ben Ford: (1)
-
- Issue and Commit Detail:
- ========================
-
- Closed Issues:
- --------------
-
- !GHSA-33x6-fj46-6rfh: Path traversal via AMI ListCategories allows access to
-                       outside files
-
- Commits By Author:
- ------------------
-
- #### Ben Ford (1):
-  - manager.c: Restrict ListCategories to the configuration directory.
-
- Commit List:
- ------------
-
- manager.c: Restrict ListCategories to the configuration directory.
-
- Commit Details:
- ---------------
-
- #### manager.c: Restrict ListCategories to the configuration directory.
-                 Author: Ben Ford
-                 Date:   2024-12-17
-
-                 When using the ListCategories AMI action, it was possible to
-                 traverse upwards through the directories to files outside of
-                 the configured configuration directory. This action is now
-                 restricted to the configured directory and an error will now be
-                 returned if the specified file is outside of this limitation.
-
-                 Resolves: #GHSA-33x6-fj46-6rfh
-
- UserNote: The ListCategories AMI action now restricts files to the configured
-           configuration directory.</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.2.0-1" date="1738843200">- The Asterisk Development Team would like to announce the release of
- asterisk-22.2.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.2.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.2.0
-
- This release resolves issues reported by the community and would
- have not been possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.2.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.2.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.1.1...22.2.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.2.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 53
- Commit Authors: 20
- Issues Resolved: 19
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### sig_analog: Add Last Number Redial feature.
-                  Users can now redial the last number called if the
-                  lastnumredial setting is set to yes.
-                  Resolves: #437
-
- #### Add SHA-256 and SHA-512-256 as authentication digest algorithms
-      The SHA-256 and SHA-512-256 algorithms are now available for
-      authentication as both a UAS and a UAC.
-
- #### Upgrade bundled pjproject to 2.15.1 Resolves: asterisk#1016
-      Bundled pjproject has been upgraded to 2.15.1. For more information visit
-      pjproject Github page:
-      https://github.com/pjsip/pjproject/releases/tag/2.15.1
-
- #### res_pjsip: Add new AOR option "qualify_2xx_only"
-                 The pjsip.conf AOR section now has a "qualify_2xx_only" option
-                 that can be set so that only 2XX responses to OPTIONS requests
-                 used to qualify a contact will mark the contact as available.
-
- #### app_queue: allow dynamically adding a queue member in paused state.
-                 Use the p option of AddQueueMember() for paused member state.
-                 Optionally, use the r(reason) option to specify a custom reason
-                 for the pause.
-
- #### manager.c: Add Processed Call Count to CoreStatus output
-                 The current processed call count is now returned as
-                 CoreProcessedCalls from the CoreStatus AMI Action.
-
- #### func_curl.c: Add additional CURL options for SSL requests
-                   The following new configuration options are now available in
-                   the res_curl.conf file, and the CURL() function:
-                   'ssl_verifyhost' (CURLOPT_SSL_VERIFYHOST), 'ssl_cainfo'
-                   (CURLOPT_CAINFO), 'ssl_capath' (CURLOPT_CAPATH), 'ssl_cert'
-                   (CURLOPT_SSLCERT), 'ssl_certtype' (CURLOPT_SSLCERTTYPE),
-                   'ssl_key' (CURLOPT_SSLKEY), 'ssl_keytype',
-                   (CURLOPT_SSLKEYTYPE) and 'ssl_keypasswd' (CURLOPT_KEYPASSWD).
-                   See the libcurl documentation for more details.
-
- #### res_stir_shaken: Allow sending Identity headers for unknown TNs
-                       You can now set the "unknown_tn_attest_level" option in
-                       the attestation and/or profile objects in
-                       stir_shaken.conf to enable sending Identity headers for
-                       callerid TNs not explicitly configured.
-
- Upgrade Notes:
- --------------
-
- #### alembic: Database updates required.
-               Two commits in this release... 'Add SHA-256 and SHA-512-256 as
-               authentication digest algorithms' 'res_pjsip: Add new AOR option
-               "qualify_2xx_only"' ...have modified alembic scripts for the
-               following database tables: ps_aors, ps_contacts, ps_auths,
-               ps_globals. If you don't use the scripts to update your database,
-               reads from those tables will succeeed but inserts into the
-               ps_contacts table by res_pjsip_registrar will fail.
-
- Commit Authors:
- ---------------
-
- Abdelkader Boudih: (3)
- Alexey Khabulyak: (1)
- Alexey Vasilyev: (1)
- Allan Nathanson: (2)
- Artem Umerov: (1)
- George Joseph: (17)
- Jaco Kroon: (1)
- James Terhune: (1)
- Joshua C. Colp: (1)
- Kent: (1)
- Maksim Nesterov: (1)
- Maximilian Fridrich: (1)
- Mike Pultz: (3)
- Naveen Albert: (6)
- Sean Bright: (6)
- Sperl Viktor: (2)
- Stanislav Abramenkov: (2)
- Steffen Arntz: (1)
- Tinet-Mucw: (1)
- Viktor Litvinov: (1)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.3.0-1" date="1743076800">- The Asterisk Development Team would like to announce the release of
- asterisk-22.3.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.3.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.3.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.3.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 28
- Commit Authors: 12
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- #### ari/pjsip: Make it possible to control transfers through ARI
-                 Call transfers on the PJSIP channel can now be controlled by
-                 ARI. This can be enabled by using the
-                 PJSIP_TRANSFER_HANDLING(ari-only) dialplan function.
-
- Commit Authors:
- ---------------
-
- Allan Nathanson: (1)
- Ben Ford: (1)
- Fabriziopicconi: (1)
- George Joseph: (10)
- Holger Hans Peter Freyther: (1)
- Jeremy Lainé: (1)
- Joshua Elson: (1)
- Luz Paz: (3)
- Maximilian Fridrich: (1)
- Mike Bradeen: (1)
- Naveen Albert: (1)
- Sean Bright: (6)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 22.4.0-1" date="1746705600">- The Asterisk Development Team would like to announce the release of
- asterisk-22.4.0.
-
- The release artifacts are available for immediate download at
- https://github.com/asterisk/asterisk/releases/tag/22.4.0 and
- https://downloads.asterisk.org/pub/telephony/asterisk
-
- Repository: https://github.com/asterisk/asterisk
- Tag: 22.4.0
-
- This release resolves issues reported by the community and would have not been
- possible without your participation.
-
- Thank You!
-
- Change Log for Release asterisk-22.4.0
- ======================================
-
- Links:
- ------
-
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.4.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.3.0...22.4.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.4.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
-
- Summary:
- --------
-
- Commits: 24
- Commit Authors: 18
- Issues Resolved: 12
- Security Advisories Resolved: 0
-
- User Notes:
- -----------
-
- stasis/control.c: Set Hangup Cause to No Answer on Dial timeout
-                   A Dial timeout on POST /channels/{channelId}/dial will now
-                   result in a CANCEL and ChannelDestroyed with cause 19 / User
-                   alerting, no answer. Previously no explicit cause was set,
-                   resulting in a cause of 16 / Normal Call Clearing.
-
- contrib: Add systemd service and timer files for malloc trim.
-          Service and timer files for systemd have been added to the
-          contrib/systemd/ directory. If you are experiencing memory issues,
-          install these files to have "malloc trim" periodically run on the
-          system.
-
- Add log-caller-id-name option to log Caller ID Name in queue log
-     This patch adds a global configuration option, log-caller-id-name, to
-     queues.conf to control whether the Caller ID name is logged as parameter 4
-     when a call enters a queue. When log-caller-id-name=yes, the Caller ID name
-     is included in the queue log, Any '|' characters in the caller ID name will
-     be replaced with '_'. (provided it’s allowed by the existing
-     log_restricted_caller_id rules). When log-caller-id-name=no (the default),
-     the Caller ID name is omitted.
-
- asterisk.c: Add "pre-init" and "pre-module" capability to cli.conf.
-             In cli.conf, you can now define startup commands that run before
-             core initialization and before module initialization.
-
- audiosocket: added support for DTMF frames
-              The AudioSocket protocol now forwards DTMF frames with payload
-              type 0x03. The payload is a 1-byte ascii representing the DTMF
-              digit (0-9,*,#...).
-
- Upgrade Notes:
- --------------
-
- ARI: REST over Websocket
-      This commit adds the ability to make ARI REST requests over the same
-      websocket used to receive events. See
-      https://docs.asterisk.org/Configuration/Interfaces/Asterisk-REST-Interface-ARI/ARI-REST-over-WebSocket/
-
- Commit Authors:
- ---------------
-
- Albrecht Oster: (1)
- Alexei Gradinari: (1)
- Allan Nathanson: (1)
- Andreas Wehrmann: (1)
- Ben Ford: (1)
- Florent CHAUVEAU: (1)
- George Joseph: (4)
- Joshua C. Colp: (1)
- Luz Paz: (1)
- Mark Murawski: (1)
- Mike Bradeen: (1)
- Mkmer: (1)
- Naveen Albert: (3)
- Norm Harrison: (2)
- Peter Jannesen: (1)
- Phoneben: (1)
- Sean Bright: (1)
- Zhai Liangliang: (1)</changelog>
</package>
<package pkgid="87931b6e1801a8779835bad9f1353c4a003daebe11920176512480d25ed1652c" name="asterisk-voicechanger" arch="src">
  <version epoch="0" ver="0.1.957ab53" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.1.957ab53" date="1769601600">- First release of this package</changelog>
</package>
<package pkgid="81d04831ee569ec5c70fa2cfb76dd2d92d4c2e6a077856d552a602840c50e97c" name="bakefile" arch="src">
  <version epoch="0" ver="0.2.9" rel="2.el7_9"/>
  <changelog author="Filipe Rosset &lt;rosset.filipe@gmail.com&gt; - 0.2.8-1" date="1261396800">- Initial RPM release</changelog>
  <changelog author="Filipe Rosset &lt;rosset.filipe@gmail.com&gt; - 0.2.8-2" date="1261483200">- Replaced summary information
- Updated BuildRequires and Requires information
- Replaced "{_mandir}/man1/bakefil*.gz" to "{_mandir}/man1/bakefil*"
- Correct URL and source URL
- Replaced "%{_libdir}/*" to "%{_libdir}/bakefile*"
- Removed useless line: "%doc"</changelog>
  <changelog author="Filipe Rosset &lt;rosset.filipe@gmail.com&gt; - 0.2.8-3" date="1267531200">- Fixed Source url
- Updated BuildRequires and Requires info
- Empy python module packaged independently
- Removed files of uuid and subprocess python modules
- Removed .la files</changelog>
  <changelog author="David Malcolm &lt;dmalcolm@redhat.com&gt; - 0.2.8-4" date="1281528000">- recompiling .py files against Python 2.7 (rhbz#623276)</changelog>
  <changelog author="Filipe Rosset &lt;rosset.filipe@gmail.com&gt; - 0.2.8-5" date="1293969600">- fix for rhbz#633520 and rhbz#652887 (thanks to Jonathan Wakely and Josef Šimánek)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.2.8-6" date="1297080000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.2.8-7" date="1326369600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.2.8-8" date="1342612800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Filipe Rosset &lt;rosset.filipe@gmail.com&gt; - 0.2.9-1" date="1352116800">- Updated to 0.2.9 upstream release</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.2.9-2" date="1360756800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
</package>
<package pkgid="5b8c73cc1c3b2d9d6aadcda646de65aa459a7a387cdcf269790307c71a2fc596" name="boost-nowide" arch="src">
  <version epoch="0" ver="0" rel="20190813.gitec9672b.el7_9"/>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 0-20171003.gitec9672b" date="1507118400">- Initial packaging</changelog>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 0-20171019.gitec9672b" date="1508414400">- Include documentation files
- Adjust to bring in line with guidelines for a header only package
- Place headers in correct place for epel</changelog>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 0-20171026.gitec9672b" date="1508414401">- Fix the directory location on EPEL to align with the boost157 package correctly
- Move the docs to a dedicated subpackage</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0-20171027.gitec9672b" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0-20171028.gitec9672b" date="1531396800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0-20171029.gitec9672b" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0-20171030.gitec9672b" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 0-20190813.gitec9672b" date="1563969601">- Simplify packaging and fix EPEL build</changelog>
</package>
<package pkgid="f69905d4cc0023cbc7c52f170eae75aa58a7f62b89388fc7d631d0b58171a9ac" name="boost159" arch="src">
  <version epoch="0" ver="1.59.0" rel="2.el7_9"/>
  <changelog author="Ha&#xEF;kel Gu&#xE9;mar &lt;hguemar@fedoraproject.org&gt; - 1.59.0-1" date="1478174400">- Initial boost159 package based on Fedora's boost-1.59.0-9</changelog>
  <changelog author="Ha&#xEF;kel Gu&#xE9;mar &lt;hguemar@fedoraproject.org&gt; - 1.59.0-2" date="1486382400">- Backport rpath patch from Fedora Rawhide
- Remove obsoletes
- Fix python-devel BR to python2-devel</changelog>
</package>
<package pkgid="8719caf2c73c643e57e4102a48056eea20581146fc00706c8dc7b21819e415d4" name="cmake3" arch="src">
  <version epoch="0" ver="3.6.1" rel="2.el7_9"/>
  <changelog author="Bj&#xF6;rn Esser &lt;fedora@besser82.io&gt; - 3.5.0-1" date="1457524800">- update to 3.5.0 final</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;fedora@besser82.io&gt; - 3.5.0-1.1" date="1457524801">- fix macros</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;fedora@besser82.io&gt; - 3.5.0-1.2" date="1457524802">- do not provide cmake = %{version}</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;fedora@besser82.io&gt; - 3.5.0-2" date="1457611200">- keep Help-directory and its contents in %_datadir/%name</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;fedora@besser82.io&gt; - 3.5.0-2.1" date="1457697600">- fix emacs-filesystem requires for epel6</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;fedora@besser82.io&gt; - 3.5.1-1" date="1458907200">- Update to 3.5.1 (#1321198)</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;fedora@besser82.io&gt; - 3.5.2-1" date="1460808000">- Update to 3.5.2 (#1327794)</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;fedora@besser82.io&gt; - 3.5.2-2" date="1461326400">- Do not own /usr/lib/rpm/fileattrs</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;fedora@besser82.io&gt; - 3.6.1-1" date="1471953600">- Update to 3.6.1 (#1353778)</changelog>
  <changelog author="Rex Dieter &lt;rdieter@fedoraproject.org&gt; 3.6.1-2" date="1472731200">- drop Provides: cmake</changelog>
</package>
<package pkgid="28a409811a4d32ae0d9163d01022e17afdb4367ca676d80337e2ba9287ccb791" name="corosync" arch="src">
  <version epoch="0" ver="3.0.0" rel="2.el7_9"/>
  <changelog author="Florian Weimer &lt;fweimer@redhat.com&gt; - 2.99.3-2" date="1532952000">- Rebuild with fixed binutils</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.3-3" date="1534766400">- Resolves: rhbz#1602409

- Remove libcgroup (rhbz#1602409)
- merge upstream commit c9e5d6db13fa965d83e27a3b664477e9b5b26edf (rhbz#1602409)</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.3-4" date="1537272000">- Related: rhbz#1615945

- Rebuild for new LibQB</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.3-5" date="1539604800">- Resolves: rhbz#1639211

- config: Fix crash in reload if new interfaces are added (rhbz#1639211)
- merge upstream commit 9f2d5a3a3faa8bd1021b505bcf3c5428b3435e39 (rhbz#1639211)</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.4-1" date="1542715200">- Related: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.4-2" date="1543924800">- Resolves: rhbz#1655179

- Add libknet1-crypto-nss-plugin dependency</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.5-1" date="1544184000">- Related: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.5-2" date="1544529600">- Resolves: rhbz#1654630

- man: Add some information about address resolution (rhbz#1654630)
- merge upstream commit 8d50bd946dd7e01da75f06da3f885e7dc82f4f12 (rhbz#1654630)
- config: Look up hostnames in a defined order (rhbz#1654630)
- merge upstream commit 3d7f136f86a56dd9d9caa9060f7a01e8b681eb7f (rhbz#1654630)</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 3.0.0-1" date="1544788800">- Resolves: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 3.0.0-2" date="1547208000">- Resolves: rhbz#1665211

- totemip: Use AF_UNSPEC for ipv4-6 and ipv6-4 (rhbz#1665211)
- merge upstream commit 2ab4d4188670356dcb82a80f2fc4598f5145c77d (rhbz#1665211)</changelog>
</package>
<package pkgid="9fb20759dfe0c3579afffc376eb71d70604aa8069e174a1dac8e1ae6772ef1ab" name="corosync" arch="src">
  <version epoch="0" ver="3.0.1" rel="1.el7_9"/>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.3-3" date="1534766400">- Resolves: rhbz#1602409

- Remove libcgroup (rhbz#1602409)
- merge upstream commit c9e5d6db13fa965d83e27a3b664477e9b5b26edf (rhbz#1602409)</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.3-4" date="1537272000">- Related: rhbz#1615945

- Rebuild for new LibQB</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.3-5" date="1539604800">- Resolves: rhbz#1639211

- config: Fix crash in reload if new interfaces are added (rhbz#1639211)
- merge upstream commit 9f2d5a3a3faa8bd1021b505bcf3c5428b3435e39 (rhbz#1639211)</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.4-1" date="1542715200">- Related: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.4-2" date="1543924800">- Resolves: rhbz#1655179

- Add libknet1-crypto-nss-plugin dependency</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.5-1" date="1544184000">- Related: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.5-2" date="1544529600">- Resolves: rhbz#1654630

- man: Add some information about address resolution (rhbz#1654630)
- merge upstream commit 8d50bd946dd7e01da75f06da3f885e7dc82f4f12 (rhbz#1654630)
- config: Look up hostnames in a defined order (rhbz#1654630)
- merge upstream commit 3d7f136f86a56dd9d9caa9060f7a01e8b681eb7f (rhbz#1654630)</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 3.0.0-1" date="1544788800">- Resolves: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 3.0.0-2" date="1547208000">- Resolves: rhbz#1665211

- totemip: Use AF_UNSPEC for ipv4-6 and ipv6-4 (rhbz#1665211)
- merge upstream commit 2ab4d4188670356dcb82a80f2fc4598f5145c77d (rhbz#1665211)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.1-1" date="1547208001">- Updated to release 3.0.1</changelog>
</package>
<package pkgid="a41b9b60ad9fddaf8671f096b584d8d5b96f4cf1763e9e5e6b308d242d5e5121" name="corosync" arch="src">
  <version epoch="0" ver="3.0.2" rel="1.el7_9"/>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.3-4" date="1537272000">- Related: rhbz#1615945

- Rebuild for new LibQB</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.3-5" date="1539604800">- Resolves: rhbz#1639211

- config: Fix crash in reload if new interfaces are added (rhbz#1639211)
- merge upstream commit 9f2d5a3a3faa8bd1021b505bcf3c5428b3435e39 (rhbz#1639211)</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.4-1" date="1542715200">- Related: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.4-2" date="1543924800">- Resolves: rhbz#1655179

- Add libknet1-crypto-nss-plugin dependency</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.5-1" date="1544184000">- Related: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.5-2" date="1544529600">- Resolves: rhbz#1654630

- man: Add some information about address resolution (rhbz#1654630)
- merge upstream commit 8d50bd946dd7e01da75f06da3f885e7dc82f4f12 (rhbz#1654630)
- config: Look up hostnames in a defined order (rhbz#1654630)
- merge upstream commit 3d7f136f86a56dd9d9caa9060f7a01e8b681eb7f (rhbz#1654630)</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 3.0.0-1" date="1544788800">- Resolves: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 3.0.0-2" date="1547208000">- Resolves: rhbz#1665211

- totemip: Use AF_UNSPEC for ipv4-6 and ipv6-4 (rhbz#1665211)
- merge upstream commit 2ab4d4188670356dcb82a80f2fc4598f5145c77d (rhbz#1665211)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.1-1" date="1547208001">- Updated to release 3.0.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.2-1" date="1560340800">- Updated to release 3.0.2</changelog>
</package>
<package pkgid="051c241a391543c8ae2a3ca592cf710311bb9c4809f67e417bcdc421ec8c7b6d" name="corosync" arch="src">
  <version epoch="0" ver="3.0.3" rel="1.el7_9"/>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.3-5" date="1539604800">- Resolves: rhbz#1639211

- config: Fix crash in reload if new interfaces are added (rhbz#1639211)
- merge upstream commit 9f2d5a3a3faa8bd1021b505bcf3c5428b3435e39 (rhbz#1639211)</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.4-1" date="1542715200">- Related: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.4-2" date="1543924800">- Resolves: rhbz#1655179

- Add libknet1-crypto-nss-plugin dependency</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.5-1" date="1544184000">- Related: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.5-2" date="1544529600">- Resolves: rhbz#1654630

- man: Add some information about address resolution (rhbz#1654630)
- merge upstream commit 8d50bd946dd7e01da75f06da3f885e7dc82f4f12 (rhbz#1654630)
- config: Look up hostnames in a defined order (rhbz#1654630)
- merge upstream commit 3d7f136f86a56dd9d9caa9060f7a01e8b681eb7f (rhbz#1654630)</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 3.0.0-1" date="1544788800">- Resolves: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 3.0.0-2" date="1547208000">- Resolves: rhbz#1665211

- totemip: Use AF_UNSPEC for ipv4-6 and ipv6-4 (rhbz#1665211)
- merge upstream commit 2ab4d4188670356dcb82a80f2fc4598f5145c77d (rhbz#1665211)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.1-1" date="1547208001">- Updated to release 3.0.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.2-1" date="1560340800">- Updated to release 3.0.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.3-1" date="1574078400">- Updated to release 3.0.3</changelog>
</package>
<package pkgid="d6811fddde4508061c56c627c1d6803ea515528336461728a8db87559f6bc673" name="corosync" arch="src">
  <version epoch="0" ver="3.0.4" rel="1.el7_9"/>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.4-1" date="1542715200">- Related: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.4-2" date="1543924800">- Resolves: rhbz#1655179

- Add libknet1-crypto-nss-plugin dependency</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.5-1" date="1544184000">- Related: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.5-2" date="1544529600">- Resolves: rhbz#1654630

- man: Add some information about address resolution (rhbz#1654630)
- merge upstream commit 8d50bd946dd7e01da75f06da3f885e7dc82f4f12 (rhbz#1654630)
- config: Look up hostnames in a defined order (rhbz#1654630)
- merge upstream commit 3d7f136f86a56dd9d9caa9060f7a01e8b681eb7f (rhbz#1654630)</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 3.0.0-1" date="1544788800">- Resolves: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 3.0.0-2" date="1547208000">- Resolves: rhbz#1665211

- totemip: Use AF_UNSPEC for ipv4-6 and ipv6-4 (rhbz#1665211)
- merge upstream commit 2ab4d4188670356dcb82a80f2fc4598f5145c77d (rhbz#1665211)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.1-1" date="1547208001">- Updated to release 3.0.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.2-1" date="1560340800">- Updated to release 3.0.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.3-1" date="1574078400">- Updated to release 3.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.4-1" date="1587556800">- Updated to release 3.0.4</changelog>
</package>
<package pkgid="25e0a5e7ef88b61f5bbf9b8aca37427e1dc3d29f28e6a1a1395befcfd314fb35" name="corosync" arch="src">
  <version epoch="0" ver="3.1.0" rel="1.el7_9"/>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.4-2" date="1543924800">- Resolves: rhbz#1655179

- Add libknet1-crypto-nss-plugin dependency</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.5-1" date="1544184000">- Related: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.5-2" date="1544529600">- Resolves: rhbz#1654630

- man: Add some information about address resolution (rhbz#1654630)
- merge upstream commit 8d50bd946dd7e01da75f06da3f885e7dc82f4f12 (rhbz#1654630)
- config: Look up hostnames in a defined order (rhbz#1654630)
- merge upstream commit 3d7f136f86a56dd9d9caa9060f7a01e8b681eb7f (rhbz#1654630)</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 3.0.0-1" date="1544788800">- Resolves: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 3.0.0-2" date="1547208000">- Resolves: rhbz#1665211

- totemip: Use AF_UNSPEC for ipv4-6 and ipv6-4 (rhbz#1665211)
- merge upstream commit 2ab4d4188670356dcb82a80f2fc4598f5145c77d (rhbz#1665211)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.1-1" date="1547208001">- Updated to release 3.0.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.2-1" date="1560340800">- Updated to release 3.0.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.3-1" date="1574078400">- Updated to release 3.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.4-1" date="1587556800">- Updated to release 3.0.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.0-1" date="1603108800">- Updated to release 3.1.0</changelog>
</package>
<package pkgid="ec265054af9931d0fb4825dd56539cac88bd4a0675877d7fdce1c3e4b1ada492" name="corosync" arch="src">
  <version epoch="0" ver="3.1.1" rel="1.el7_9"/>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 2.99.5-1" date="1544184000">- Related: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.5-2" date="1544529600">- Resolves: rhbz#1654630

- man: Add some information about address resolution (rhbz#1654630)
- merge upstream commit 8d50bd946dd7e01da75f06da3f885e7dc82f4f12 (rhbz#1654630)
- config: Look up hostnames in a defined order (rhbz#1654630)
- merge upstream commit 3d7f136f86a56dd9d9caa9060f7a01e8b681eb7f (rhbz#1654630)</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 3.0.0-1" date="1544788800">- Resolves: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 3.0.0-2" date="1547208000">- Resolves: rhbz#1665211

- totemip: Use AF_UNSPEC for ipv4-6 and ipv6-4 (rhbz#1665211)
- merge upstream commit 2ab4d4188670356dcb82a80f2fc4598f5145c77d (rhbz#1665211)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.1-1" date="1547208001">- Updated to release 3.0.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.2-1" date="1560340800">- Updated to release 3.0.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.3-1" date="1574078400">- Updated to release 3.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.4-1" date="1587556800">- Updated to release 3.0.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.0-1" date="1603108800">- Updated to release 3.1.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.1-1" date="1617019200">- Updated to release 3.1.1</changelog>
</package>
<package pkgid="7356f7ab5113140434d87c2af5bb6d66b4d6daf825180b8005a987e14b0e7001" name="corosync" arch="src">
  <version epoch="0" ver="3.1.2" rel="1.el7_9"/>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 2.99.5-2" date="1544529600">- Resolves: rhbz#1654630

- man: Add some information about address resolution (rhbz#1654630)
- merge upstream commit 8d50bd946dd7e01da75f06da3f885e7dc82f4f12 (rhbz#1654630)
- config: Look up hostnames in a defined order (rhbz#1654630)
- merge upstream commit 3d7f136f86a56dd9d9caa9060f7a01e8b681eb7f (rhbz#1654630)</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 3.0.0-1" date="1544788800">- Resolves: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 3.0.0-2" date="1547208000">- Resolves: rhbz#1665211

- totemip: Use AF_UNSPEC for ipv4-6 and ipv6-4 (rhbz#1665211)
- merge upstream commit 2ab4d4188670356dcb82a80f2fc4598f5145c77d (rhbz#1665211)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.1-1" date="1547208001">- Updated to release 3.0.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.2-1" date="1560340800">- Updated to release 3.0.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.3-1" date="1574078400">- Updated to release 3.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.4-1" date="1587556800">- Updated to release 3.0.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.0-1" date="1603108800">- Updated to release 3.1.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.1-1" date="1617019200">- Updated to release 3.1.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.2-1" date="1617710400">- Updated to release 3.1.2</changelog>
</package>
<package pkgid="437429d21c593ca0f8b7577d8861c20001e9700a12d224c99654992c6a5ea812" name="corosync" arch="src">
  <version epoch="0" ver="3.1.3" rel="1.el7_9"/>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; - 3.0.0-1" date="1544788800">- Resolves: rhbz#1600915

- New upstream release</changelog>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 3.0.0-2" date="1547208000">- Resolves: rhbz#1665211

- totemip: Use AF_UNSPEC for ipv4-6 and ipv6-4 (rhbz#1665211)
- merge upstream commit 2ab4d4188670356dcb82a80f2fc4598f5145c77d (rhbz#1665211)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.1-1" date="1547208001">- Updated to release 3.0.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.2-1" date="1560340800">- Updated to release 3.0.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.3-1" date="1574078400">- Updated to release 3.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.4-1" date="1587556800">- Updated to release 3.0.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.0-1" date="1603108800">- Updated to release 3.1.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.1-1" date="1617019200">- Updated to release 3.1.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.2-1" date="1617710400">- Updated to release 3.1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.3-1" date="1621598400">- Updated to release 3.1.3</changelog>
</package>
<package pkgid="0ae81d34cef58bbf6580405178dc9a0634db0723834c716434e04c3d6bbdfa35" name="corosync" arch="src">
  <version epoch="0" ver="3.1.4" rel="1.el7_9"/>
  <changelog author="Jan Friesse &lt;jfriesse@redhat.com&gt; 3.0.0-2" date="1547208000">- Resolves: rhbz#1665211

- totemip: Use AF_UNSPEC for ipv4-6 and ipv6-4 (rhbz#1665211)
- merge upstream commit 2ab4d4188670356dcb82a80f2fc4598f5145c77d (rhbz#1665211)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.1-1" date="1547208001">- Updated to release 3.0.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.2-1" date="1560340800">- Updated to release 3.0.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.3-1" date="1574078400">- Updated to release 3.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.4-1" date="1587556800">- Updated to release 3.0.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.0-1" date="1603108800">- Updated to release 3.1.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.1-1" date="1617019200">- Updated to release 3.1.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.2-1" date="1617710400">- Updated to release 3.1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.3-1" date="1621598400">- Updated to release 3.1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.4-1" date="1622721600">- Updated to release 3.1.4</changelog>
</package>
<package pkgid="413fd4ce2b6e38169309a25862f65e3637f2811bb180d663ea63713985778114" name="corosync" arch="src">
  <version epoch="0" ver="3.1.5" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.1-1" date="1547208000">- Updated to release 3.0.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.2-1" date="1560340800">- Updated to release 3.0.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.3-1" date="1574078400">- Updated to release 3.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.4-1" date="1587556800">- Updated to release 3.0.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.0-1" date="1603108800">- Updated to release 3.1.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.1-1" date="1617019200">- Updated to release 3.1.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.2-1" date="1617710400">- Updated to release 3.1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.3-1" date="1621598400">- Updated to release 3.1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.4-1" date="1622721600">- Updated to release 3.1.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.5-1" date="1628078400">- Updated to release 3.1.5</changelog>
</package>
<package pkgid="dad15d78b04076ed8906bdb8aa4b381cf95d5ac284fbad838a01ff59cd391d5d" name="corosync" arch="src">
  <version epoch="0" ver="3.1.6" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.2-1" date="1560340800">- Updated to release 3.0.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.3-1" date="1574078400">- Updated to release 3.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.4-1" date="1587556800">- Updated to release 3.0.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.0-1" date="1603108800">- Updated to release 3.1.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.1-1" date="1617019200">- Updated to release 3.1.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.2-1" date="1617710400">- Updated to release 3.1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.3-1" date="1621598400">- Updated to release 3.1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.4-1" date="1622721600">- Updated to release 3.1.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.5-1" date="1628078400">- Updated to release 3.1.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.6-1" date="1636545600">- Updated to release 3.1.6</changelog>
</package>
<package pkgid="660db01bb097f029a280533716b11ec3101c92d814458ba2dd8d461fd7845b6d" name="corosync" arch="src">
  <version epoch="0" ver="3.1.7" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.3-1" date="1574078400">- Updated to release 3.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.4-1" date="1587556800">- Updated to release 3.0.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.0-1" date="1603108800">- Updated to release 3.1.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.1-1" date="1617019200">- Updated to release 3.1.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.2-1" date="1617710400">- Updated to release 3.1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.3-1" date="1621598400">- Updated to release 3.1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.4-1" date="1622721600">- Updated to release 3.1.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.5-1" date="1628078400">- Updated to release 3.1.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.6-1" date="1636545600">- Updated to release 3.1.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.7-1" date="1666612800">- Updated to release 3.1.7</changelog>
</package>
<package pkgid="25b89310aab0ebb6e2fa504947672e0456bbe760a4d08e8d609c03ffa14a6697" name="coturn" arch="src">
  <version epoch="0" ver="4.5.1.1" rel="1.el7_9"/>
  <changelog author="Christian Glombek &lt;lorbus@fedoraproject.org&gt; - 4.5.1.0-1" date="1550491200">- Initial Fedora Package
- Update to 4.5.1.0
- Introduce consistent naming, rename service to coturn
- Add configure, make and systemd macros
- Remove dependencies on mariadb, mysql, postgresql and sqlite
- Forked from https://github.com/coturn/coturn/blob/af674368d120361603342ff4ff30b44f147a38ff/rpm/turnserver.spec</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.5.1.0-2" date="1563969600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Robert Scheck &lt;robert@fedoraproject.org&gt; - 4.5.1.0-3" date="1564142400">- Added patch to append only to log files rather to override always
- Relocate SQLite database to FHS conform /var/lib/coturn/turndb path
- Include default log file directory with logrotate configuration
- Provide /run/coturn and correct PID file handling (#1705146)
- Ensure private keys for SSL certs can be only read by coturn user
- Ensure /etc/coturn/turnserver.conf can be only read by coturn user
- Correct subpackage licensing as per Fedora Packaging Guidelines</changelog>
  <changelog author="Robert Scheck &lt;robert@fedoraproject.org&gt; - 4.5.1.1-1" date="1564228800">- Update to 4.5.1.1</changelog>
</package>
<package pkgid="fc8c3644fe8399194f972e078f5c9eabac0af13ec8f8ccdbee3de8f83bac4e3e" name="cpp-hocon" arch="src">
  <version epoch="0" ver="0.1.6" rel="9.el7_9"/>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 0.1.6-1" date="1507118400">- Initial packaging</changelog>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 0.1.6-2" date="1508414400">- rebuilt</changelog>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 0.1.6-3" date="1508932800">- Point to correct location on epel7 for leatherman cmake3 files</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 0.1.6-4" date="1516708800">- Rebuilt for Boost 1.66</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.6-5" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Filipe Rosset &lt;rosset.filipe@gmail.com&gt; - 0.1.6-6" date="1518696000">- fix FTBFS</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 0.1.6-7" date="1519387200">- Disable tests (fails on x86)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.6-8" date="1531396800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.6-9" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
</package>
<package pkgid="07cc9fcbced205855af697a2f40775e85088b95c6775b0f0ee46300cd6770ea6" name="dahdi-tools" arch="src">
  <version epoch="0" ver="2.10.2" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.4.1-9" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Bruno Wolff III &lt;bruno@wolff.to&gt; - 2.7.0-1" date="1376740800">- Move to current upstream release
- Changlog is available at http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog</changelog>
  <changelog author="Bruno Wolff III &lt;bruno@wolff.to&gt; - 2.7.0-2" date="1376740801">- Include user.h as dahdi_config.h to avoid issue noted in bug 783890</changelog>
  <changelog author="Bruno Wolff III &lt;bruno@wolff.to&gt; - 2.7.0-3" date="1376740802">- blacklist wcopenpci so that it doesn't get loaded until the right time</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.7.0-4" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.7.0-5" date="1408190400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 2.7.0-6" date="1409140800">- Perl 5.20 rebuild</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 2.10.0-1" date="1414152000">- Update to 2.10.0</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 2.10.0-2" date="1415361600">- Build-require perl-podlators for pod2man tool (bug #1161158)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.10.2-1" date="1504180800">- Update to 2.10.2
- Added Aligera AP200 and AP400 reference at module file
- Added fxstest tool build</changelog>
</package>
<package pkgid="d4ffb06a0492ad96356a530e8e60d5a980a62296b5cdb5f55ded33f81f5306db" name="dahdi-tools" arch="src">
  <version epoch="0" ver="2.10.2" rel="2.el7_9"/>
  <changelog author="Bruno Wolff III &lt;bruno@wolff.to&gt; - 2.7.0-1" date="1376740800">- Move to current upstream release
- Changlog is available at http://git.asterisk.org/gitweb/?p=dahdi/tools.git;a=shortlog</changelog>
  <changelog author="Bruno Wolff III &lt;bruno@wolff.to&gt; - 2.7.0-2" date="1376740801">- Include user.h as dahdi_config.h to avoid issue noted in bug 783890</changelog>
  <changelog author="Bruno Wolff III &lt;bruno@wolff.to&gt; - 2.7.0-3" date="1376740802">- blacklist wcopenpci so that it doesn't get loaded until the right time</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.7.0-4" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.7.0-5" date="1408190400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 2.7.0-6" date="1409140800">- Perl 5.20 rebuild</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 2.10.0-1" date="1414152000">- Update to 2.10.0</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 2.10.0-2" date="1415361600">- Build-require perl-podlators for pod2man tool (bug #1161158)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.10.2-1" date="1504180800">- Update to 2.10.2
- Added Aligera AP200 and AP400 reference at module file
- Added fxstest tool build</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.10.2-2" date="1542110400">- Updated Aligera patch to revision 33</changelog>
</package>
<package pkgid="cca2f85e17d53119bc460833670f434927c5fced6bca6283a9d5a8a0e82a0562" name="dhcp_probe" arch="src">
  <version epoch="0" ver="1.3.0" rel="18.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3.0-9" date="1326456000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Jon Ciesla &lt;limburgher@gmail.com&gt; - 1.3.0-10" date="1333108800">- libnet rebuild.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3.0-11" date="1342612800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Michael Schwendt &lt;mschwendt@fedoraproject.org&gt; - 1.3.0-12" date="1344340800">- Fix for 64-bit architecture (#727308).</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3.0-13" date="1360756800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3.0-14" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3.0-15" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3.0-16" date="1408190400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3.0-17" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.3.0-18" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
</package>
<package pkgid="e347700b924597d3f9435471776e2a64d69b7bb9a06e5cd7bfb5236e0cfdeb3a" name="docbook2X" arch="src">
  <version epoch="0" ver="0.8.8" rel="28.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.8.8-19" date="1408190400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 0.8.8-20" date="1409227200">- Perl 5.20 rebuild</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 0.8.8-21" date="1433592000">- Perl 5.22 rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.8.8-22" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.8.8-23" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 0.8.8-24" date="1463400000">- Perl 5.24 rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.8.8-25" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 0.8.8-26" date="1496664000">- Perl 5.26 rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.8.8-27" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.8.8-28" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
</package>
<package pkgid="2757c15bccb98053a567c5fdc32bb5f9fb748dba6105a6c9d651f6ccd79cd3c7" name="enca" arch="src">
  <version epoch="0" ver="1.19" rel="1.el7_9"/>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.10-1" date="1251201600">- Update to 1.10
- Change urls for new upstream</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.13-1" date="1269864000">- update to 1.13</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.13-2" date="1297166400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.13-3" date="1326456000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.13-4" date="1342612800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.13-5" date="1360756800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.14-1" date="1362484800">- update to 1.14</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.16-1" date="1441886400">- update to 1.16</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.18-1" date="1455192000">- update to 1.18</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 1.19-1" date="1522238400">- update to 1.19
- spec file cleanups
- avoid rpath in binaries</changelog>
</package>
<package pkgid="396c86fbb9d260f24f19e74e21f279ef9dfe60576777afe6326209d055c85a20" name="faac" arch="src">
  <version epoch="0" ver="1.28" rel="6.0.el7_9"/>
  <changelog author="Thorsten Leemhuis &lt;fedora [AT] leemhuis [DOT] info - 1.25-5" date="1216728000">- rebuild for RPM Fusion</changelog>
  <changelog author="Thorsten Leemhuis &lt;fedora [AT] leemhuis [DOT] info - 1.25-6" date="1216900800">- rebuild</changelog>
  <changelog author="Thorsten Leemhuis &lt;fedora [AT] leemhuis [DOT] info - 1.25-7" date="1225800000">- chmod 644 all docs (fixes #115)</changelog>
  <changelog author="Thorsten Leemhuis &lt;fedora[AT]leemhuis.info&gt; - 1.26-1" date="1229256000">- ship AUTHORS NEWS docs/*
- integrated changes from Julian Sikorski &lt;belegdol[at]gmail[dot]com&gt;
-- Updated to 1.26
-- Dropped dos2unix BR, not needed anymore
-- Made Source0 use macros
-- Fixed License tag
-- Fixed file permissions
-- Converted ChangeLog to utf-8</changelog>
  <changelog author="Dominik Mierzejewski &lt;rpm [AT] greysector [DOT] net&gt; - 1.28-1" date="1236859200">- update to 1.28
- drop redundant BRs
- fix Source URL and use bz2 tarball
- fix rpath
- include manpage</changelog>
  <changelog author="Thorsten Leemhuis &lt;fedora [AT] leemhuis [DOT] info&gt; - 1.28-1.1" date="1239105600">- rebuild</changelog>
  <changelog author="kwizart &lt; kwizart at gmail.com &gt; - 1.28-2" date="1256472000">- Install with -p
- Moved in nonfree</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 1.28-4" date="1328788800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 1.28-5" date="1363089600">- https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Nux &lt;rpm@li.nux.ro&gt; - 1.28-6" date="1422619200">- update to build against newer libmp4v2, thanks Constantin
- https://forums.nux.ro/index.php?t=msg&amp;goto=2996</changelog>
</package>
<package pkgid="eff2078e2a50d2e0991c8a50112b2563903b1aad42ad853aeaccb1372ef7d751" name="faad2" arch="src">
  <version epoch="1" ver="2.7" rel="5.el7_9"/>
  <changelog author="Dominik Mierzejewski &lt;dominik [AT] greysector [DOT] net&gt; 1:2.6.1-2" date="1194782400">- bring back the XMMS plugin
- move EOL fixup to prep
- fix URLs</changelog>
  <changelog author="Dominik Mierzejewski &lt;dominik [AT] greysector [DOT] net&gt; 1:2.6.1-3" date="1200225600">- split off libs to avoid multilib conflicts</changelog>
  <changelog author="Thorsten Leemhuis &lt;fedora [AT] leemhuis [DOT] info - 1:2.6.1-4" date="1217764800">- rebuild</changelog>
  <changelog author="Dominik Mierzejewski &lt;dominik [AT] greysector [DOT] net&gt; 1:2.6.1-5" date="1224331200">- add Obsoletes: for xmms-aac to ensure smooth upgrade from Freshrpms
- add some additional docs for xmms-faad2</changelog>
  <changelog author="Dominik Mierzejewski &lt;dominik [AT] greysector [DOT] net&gt; 1:2.6.1-6" date="1226318400">- fix CVE-2008-4201</changelog>
  <changelog author="Dominik Mierzejewski &lt;dominik [AT] greysector [DOT] net&gt; 1:2.7-1" date="1236945600">- update to 2.7
- don't install internal libmp4ff
- include manpage
- fix build on x86_64 (non-PIC objects in libmp4ff.a)
- fix rpaths
- make xmms plugin depend on -libs, not the frontend
- preserve docs timestamps</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 1:2.7-2" date="1327492800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 1:2.7-3" date="1362312000">- Mass rebuilt for Fedora 19 Features</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 1:2.7-4" date="1364385600">- Change the escaping space hack - rhbz#928110</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 1:2.7-5" date="1388404800">- Rebuilt for F-20</changelog>
</package>
<package pkgid="235f94efb540b1628c9f7d80db233718c0754628c601dd4ae1af267db50e9881" name="facter" arch="src">
  <version epoch="1" ver="3.9.3" rel="7.el7_9"/>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 3.9.2-1" date="1507118400">- Update to latest upstream version 3.9.2</changelog>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 3.9.2-2" date="1508414400">- rebuilt</changelog>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 3.9.2-3" date="1508932800">- Point to correct leatherman directory on cmake3 for epel7</changelog>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 3.9.3-1" date="1510056000">- new upstream release 3.9.3</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 3.9.3-2" date="1516708800">- Rebuilt for Boost 1.66</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.9.3-3" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Richard Shaw &lt;hobbes1069@gmail.com&gt; - 3.9.3-4" date="1518609600">- Rebuild for yaml-cpp 0.6.0.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.9.3-5" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.9.3-6" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Alfredo Moralejo &lt;amoralej@redhat.com&gt; - 3.9.3-7" date="1561032000">- Fix bug when using epoch for RDO build.</changelog>
</package>
<package pkgid="de18e30f92cc0f64ea84499d0a5a2bab4200a2be02675588df172d9c58c14290" name="fdk-aac" arch="src">
  <version epoch="0" ver="0.1.4" rel="1.el7_9"/>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.1-1" date="1364472000">- Initial spec</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.2-1" date="1376568000">- Update to 0.1.2</changelog>
  <changelog author="Nux &lt;rpm@li.nux.ro&gt; - 0.1.4-1" date="1458648000">- update to 0.1.4</changelog>
</package>
<package pkgid="234d753733749cca321fa8816c032813a9603a808c7f3c6f62c3eb1e07202522" name="ffmpeg" arch="src">
  <version epoch="0" ver="2.6.8" rel="3.el7_9"/>
  <changelog author="Dominik Mierzejewski &lt;rpm@greysector.net&gt; - 2.4.8-3" date="1429099200">- rebuilt for new x265</changelog>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.4.9-1" date="1430222400">- Updated to 2.4.9</changelog>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.6.2-1" date="1430827200">- Updated to 2.6.2</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.6.2-2" date="1431345600">- Disable opencl by default - rfbz#3640
- Add with condition for nvenc,fdk_aac</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.6.2-3" date="1431777600">- Rebuilt for x265</changelog>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.6.3-1" date="1432728000">- Updated to 2.6.3</changelog>
  <changelog author="Nux &lt;rpm@li.nux.ro&gt; - 2.6.5-1" date="1449057600">- update to 2.6.5</changelog>
  <changelog author="Gordan Bobic &lt;gordan@redsleeve.org&gt; - 2.6.8-1" date="1455710400">- Update to 2.6.8
- Enable nvenc
- Enable libfdk-aac
- Enable libfaac</changelog>
  <changelog author="Nux &lt;rpm@li.nux.ro&gt; - 2.6.8-1" date="1458648000">- disable libvpx - EL7's version is too old to support ffmpeg</changelog>
  <changelog author="Nux &lt;rpm@li.nux.ro&gt; - 2.6.8-3" date="1461758400">- recent nvenc enhancement breaks conversions, undoing</changelog>
</package>
<package pkgid="1a4e0011ac389f4eafb3f8c12e79522969b4fd444cc0cd25b688386bdd041540" name="ffmpeg" arch="src">
  <version epoch="0" ver="2.8.14" rel="1.el7_9"/>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.8.6-1" date="1454328000">- Updated to 2.8.6</changelog>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.8.7-1" date="1462190400">- Updated to 2.8.7</changelog>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.8.8-1" date="1474459200">- Updated to 2.8.8</changelog>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.8.9-1" date="1480852800">- Updated to 2.8.9</changelog>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.8.10-1" date="1481025600">- Updated to 2.8.10
- Backported some .spec cleanups from rawhide</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.11-1" date="1486987200">- Update to 2.8.11
- enable AMR codecs by default - rfbz#4367</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.12-1" date="1496836800">- Update to 2.8.12</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.13-1" date="1504353600">- Update to 2.8.13 (security issues)</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.13-2" date="1517313600">- rebuilt</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.14-1" date="1519300800">- Update to 2.8.14</changelog>
</package>
<package pkgid="304dbd668248de07fc5f800f0bb02e506431d79b592405ed1f8432057171b9fe" name="ffmpeg" arch="src">
  <version epoch="0" ver="2.8.15" rel="1.el7_9"/>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.8.7-1" date="1462190400">- Updated to 2.8.7</changelog>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.8.8-1" date="1474459200">- Updated to 2.8.8</changelog>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.8.9-1" date="1480852800">- Updated to 2.8.9</changelog>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.8.10-1" date="1481025600">- Updated to 2.8.10
- Backported some .spec cleanups from rawhide</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.11-1" date="1486987200">- Update to 2.8.11
- enable AMR codecs by default - rfbz#4367</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.12-1" date="1496836800">- Update to 2.8.12</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.13-1" date="1504353600">- Update to 2.8.13 (security issues)</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.13-2" date="1517313600">- rebuilt</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.14-1" date="1519300800">- Update to 2.8.14</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 2.8.15-1" date="1531915200">- Update to 2.8.15</changelog>
</package>
<package pkgid="70b580b17632d5e0b7c45111f4b24b028c39373cf66b01edf8dfc5d61ca7c7af" name="ffmpeg" arch="src">
  <version epoch="0" ver="2.8.18" rel="1.el7_9"/>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.8.8-1" date="1474459200">- Updated to 2.8.8</changelog>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.8.9-1" date="1480852800">- Updated to 2.8.9</changelog>
  <changelog author="Julian Sikorski &lt;belegdol@fedoraproject.org&gt; - 2.8.10-1" date="1481025600">- Updated to 2.8.10
- Backported some .spec cleanups from rawhide</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.11-1" date="1486987200">- Update to 2.8.11
- enable AMR codecs by default - rfbz#4367</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.12-1" date="1496836800">- Update to 2.8.12</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.13-1" date="1504353600">- Update to 2.8.13 (security issues)</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.13-2" date="1517313600">- rebuilt</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 2.8.14-1" date="1519300800">- Update to 2.8.14</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 2.8.15-1" date="1531915200">- Update to 2.8.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.8.18-1" date="1635336000">- Update to 2.8.18</changelog>
</package>
<package pkgid="6106ab05e6835d28d0f524da8cec3c0fe0c25d3f809c31874b33178d03ea9fc5" name="firebird" arch="src">
  <version epoch="0" ver="2.5.5.26952.0" rel="2.el7_9"/>
  <changelog author="Philippe Makowski &lt;makowski@fedoraproject.org&gt;  2.5.2.26539.0-4" date="1366891200">- set PIE compiler flags (bug #955274)</changelog>
  <changelog author="Philippe Makowski &lt;makowski@fedoraproject.org&gt;  2.5.2.26539.0-5" date="1370260800">- Firebird fails to build for aarch64  (bug #969851)</changelog>
  <changelog author="Philippe Makowski &lt;makowski@fedoraproject.org&gt;  2.5.2.26539.0-6" date="1374580800">- Provide fb_config in firebird-devel  (bug #985335)</changelog>
  <changelog author="Philippe Makowski &lt;makowski@fedoraproject.org&gt;  2.5.2.26539.0-7" date="1374580801">- make fb_config executable  (bug #985335)</changelog>
  <changelog author="Philippe Makowski &lt;makowski@fedoraproject.org&gt;  2.5.2.26539.0-8" date="1390305600">- add missing BR on libstdc++-static</changelog>
  <changelog author="Philippe Makowski &lt;makowski@fedoraproject.org&gt;  - 2.5.3.26778.0-1" date="1406376000">- update from upstream 2.5.3
- update arm64 patch</changelog>
  <changelog author="Philippe Makowski &lt;makowski@fedoraproject.org&gt; 2.5.3.26778.0-2" date="1417953600">- security fix firebird CORE-4630</changelog>
  <changelog author="Philippe Makowski &lt;makowski@fedoraproject.org&gt; 2.5.4.26856.0-1" date="1427976000">- update to 2.5.4</changelog>
  <changelog author="Philippe Makowski &lt;makowski@fedoraproject.org&gt; 2.5.5.26952.0-1" date="1447934400">- update to 2.5.5</changelog>
  <changelog author="Philippe Makowski &lt;makowski@fedoraproject.org&gt; - 2.5.5.26952.0-2" date="1454673600">- move fb_config (#1297506)
- fix CVE-2016-1569 (#1297447 #1297450 #1297451)</changelog>
</package>
<package pkgid="323e52b13bb6e792abdf4c72bbd0c8993a155f3aef9c0db6d249ccf29029f82a" name="freelan" arch="src">
  <version epoch="0" ver="2.2" rel="1.el7_9"/>
  <changelog author="Jethro Carr &lt;jethro.carr@jethrocarr.com&gt; - 2.0.20150131git-1" date="1422705600">- Initial Build</changelog>
  <changelog author="Jethro Carr &lt;jethro.carr@jethrocarr.com&gt; - 2.0.20150131git-2" date="1424952000">- Fixed a mistake in the systemd init file</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2-1" date="1654171200">- Updated to release 2.2</changelog>
</package>
<package pkgid="fbea96a6de92bd6be0618620e86acbab9b4d2c0e5677057d3b96832bd7479c5f" name="freeradius" arch="src">
  <version epoch="0" ver="3.0.13" rel="6.el7_9"/>
  <changelog author="Nikolai Kondrashov &lt;Nikolai.Kondrashov@redhat.com&gt; - 3.0.10-1" date="1449662400">- Upgrade to upstream v3.0.10 release.
  See upstream ChangeLog for details (in freeradius-doc subpackage).
  Related: Bug#1202751
  Resolves: Bug#1340334 freeradius: Decryption of very long Tunnel-Passwords
                        can cause buffer overflow
- Remove rlm_eap_tnc support as the required package "tncfhh" was retired.</changelog>
  <changelog author="Nikolai Kondrashov &lt;Nikolai.Kondrashov@redhat.com&gt; - 3.0.11-1" date="1460980800">- Upgrade to upstream v3.0.11 release.
  See upstream ChangeLog for details (in freeradius-doc subpackage).
  Resolves: Bug#1197551
  Resolves: Bug#1179745
  Resolves: Bug#1202751 Rebase FreeRADIUS to 3.0.7 or later minor release
  Resolves: Bug#1289849 FreeRadius should start after ldap, ipa and krb5kdc
  Resolves: Bug#1208886 Add the latest Mikrotik dictionary into Freeradius
  Resolves: Bug#1198620 radutmp should not rotate
  Resolves: Bug#1180979 Freeradius is installing files under /etc/tmpfiles.d/
  Resolves: Bug#1187904 radiusd logrotate config file contains old style
                        "/sbin/service radius reload" reload call
  Resolves: Bug#1167846 radiusd fails to load clients from ldap
  Resolves: Bug#1422018 /usr/lib/systemd/system/radiusd.service is marked
                        executable. Please remove executable permission bits
  Resolves: Bug#1167843 support for older style generic attributes in
                        rlm_ldap doesn't work
  Resolves: Bug#1354234 home servers are marked as dead by radiusd</changelog>
  <changelog author="Nikolai Kondrashov &lt;Nikolai.Kondrashov@redhat.com&gt; - 3.0.12-1" date="1480075200">- Upgrade to upstream v3.0.12 release.
  See upstream ChangeLog for details (in freeradius-doc subpackage).
  Related:  Bug#1202751 Rebase FreeRADIUS to 3.0.7 or later minor release
  Resolves: Bug#1358989 FreeRADIUS stops reading the accounting packets from the
                        detail file
  Resolves: Bug#1269217 FreeRADIUS triggers mprotect with large radius reply
  Resolves: Bug#1344183 radiusd crashed in rbtree_find() after receiving a packet
  Resolves: Bug#1370431 FreeRADIUS fails to establish LDAP connections under
                        load
  Resolves: Bug#1397981 [abrt] freeradius: radiusd killed by SIGSEGV</changelog>
  <changelog author="Nikolai Kondrashov &lt;Nikolai.Kondrashov@redhat.com&gt; - 3.0.12-2" date="1487678400">- Do not fail logrotate if radiusd is not running.
  Resolves: Bug#1365226 error running non-shared postrotate script for
                        /var/log/radius/radius.log of
                        '/var/log/radius/radius.log '
- Fix output to log file specified with -l option.
  Resolves: Bug#1421207 radiusd does not work with log file specified by -l
                        option
- Fix long hostnames interpreted as IP addresses.
  Resolves: Bug#1420359 radclient does not detect 4. level and higher domain
                        name as a domain name
- Avoid clashes with libtool library symbols.
  Resolves: Bug#1391960 undefined symbol: get_vtable in
                        /usr/lib64/libtdsodbc.so.0 with freeradius-unixODBC
- Remove mentions of Auth-Type = System from docs.
  Resolves: Bug#1420293 Freeradius does not know Auth-Type = System
- Improve ip/v4/v6/addr documentation.
  Resolves: Bug#1179736 add description for ipaddr = hostname from DNS with A
                        and AAAA entries</changelog>
  <changelog author="Nikolai Kondrashov &lt;Nikolai.Kondrashov@redhat.com&gt; - 3.0.13-1" date="1488888000">- Upgrade to upstream v3.0.13 release.
  See upstream ChangeLog for details (in freeradius-doc subpackage).
  Related:  Bug#1202751 Rebase FreeRADIUS to 3.0.12 or later minor release
  Resolves: Bug#1329181 freeradius-python fix libpython2.7.so dependency at
                        compile time
  Resolves: Bug#1425869 Radiusd does not trigger modules.sql.fail trap if it
                        can't connect to mysql server
  Resolves: Bug#1427829 Stack overflow when passing bare IPv6 to radclient</changelog>
  <changelog author="Nikolai Kondrashov &lt;Nikolai.Kondrashov@redhat.com&gt; - 3.0.13-2" date="1490270400">- Fix some issues found with static analyzers.
  Resolves: Bug#1432103 FreeRADIUS fails covscan checks
- Revert rlm_eap_tnc removal, because the tncfhh package is still in RHEL.
  Related:  Bug#1202751 Rebase FreeRADIUS to 3.0.12 or later minor release</changelog>
  <changelog author="Nikolai Kondrashov &lt;Nikolai.Kondrashov@redhat.com&gt; - 3.0.13-3" date="1490788800">- Explicitly disable rlm_cache_memcached to avoid error when the module's
  dependencies are installed, and it is built, but not packaged.
  Related:  Bug#1202751 Rebase FreeRADIUS to 3.0.12 or later minor release
- Prevent segfaults by adding a missing handling of connection errors in
  rlm_ldap.
  Resolves: Bug#1437409 [abrt] freeradius: radiusd killed by SIGSEGV
- Make radtest use Cleartext-Password for EAP, fixing its support for eap-md5.
  Resolves: Bug#1436619 radtest does not work with eap-md5</changelog>
  <changelog author="Nikolai Kondrashov &lt;Nikolai.Kondrashov@redhat.com&gt; - 3.0.13-4" date="1496750400">- Disable internal OpenSSL cache.
  Resolves: Bug#1458746 CVE-2017-9148 freeradius: TLS resumption
                        authentication bypass
- Check sizeof(*packet), not sizeof(packet) in EAP-PWD
  Resolves: Bug#1459073 Potential buffer overflow in FreeRADIUS EAP-PWD
- Parse port as well as address for dhcpclient destination
  Resolves: Bug#1457825 dhcpclient does no accept IP:PORT</changelog>
  <changelog author="Nikolai Kondrashov &lt;Nikolai.Kondrashov@redhat.com&gt; - 3.0.13-5" date="1496750401">- Refer to correct package version in configuration comments for Bug#1458746
  (CVE-2017-9148) fix.
  Related: Bug#1458746  CVE-2017-9148 freeradius: TLS resumption
                        authentication bypass</changelog>
  <changelog author="Nikolai Kondrashov &lt;Nikolai.Kondrashov@redhat.com&gt; - 3.0.13-6" date="1497528000">- Avoid race condition when creating session cache file
  Resolves: Bug#1458746 CVE-2017-9148 freeradius: TLS resumption
                        authentication bypass</changelog>
</package>
<package pkgid="1466c398888904e94c383f92c3bccfbf5b58be5f9e0f5cb43a153c4810a6cee2" name="freeradius-client" arch="src">
  <version epoch="0" ver="1.1.7" rel="7.el7_9"/>
  <changelog author="Nikos Mavrogiannopoulos - 1.1.6-1" date="1417780800">- Initial package release</changelog>
  <changelog author="Nikos Mavrogiannopoulos - 1.1.6.20141208-1.gite261681" date="1418040000">- Rebase on git repository</changelog>
  <changelog author="Nikos Mavrogiannopoulos - 1.1.7-1" date="1421928000">- New upstream release</changelog>
  <changelog author="Nikos Mavrogiannopoulos - 1.1.7-2" date="1422360000">- Cleanup licensing
- Link to main upstream web page
- Properly obsolete radiusclient-ng-utils
- Remove dependencies on autotools</changelog>
  <changelog author="Nikos Mavrogiannopoulos - 1.1.7-3" date="1422446400">- Line wrapped description message
- Commented out the utils subpackage</changelog>
  <changelog author="Nikos Mavrogiannopoulos - 1.1.7-4" date="1424692800">- Corrected paths of files includes in config file</changelog>
  <changelog author="Nikos Mavrogiannopoulos - 1.1.7-6" date="1427198400">- when sending IPv6 attributes use the correct length (#1205156)</changelog>
  <changelog author="Nikos Mavrogiannopoulos - 1.1.7-7" date="1433764800">- Ship a custom dictionary; that corrects the IPv6 dictionary types.</changelog>
</package>
<package pkgid="be3188512aa765481f1d92b88015be9e4b70904caef4cad97fff5564d7f99651" name="freetds" arch="src">
  <version epoch="0" ver="0.91" rel="3.el7_9"/>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 0.82-4" date="1235476800">- fix autoconf data for libtool2 (patch by Tom Lane &lt;tgl@redhat.com&gt;)</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 0.82-5" date="1238068800">- add upstream patch cspublic.BLK_VERSION_150.patch (#492393)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.82-6" date="1248436800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.82-7" date="1297166400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 0.82.1-0.1.20100810dev" date="1297339200">- update to the latest stable snapshot 0.82.1.dev.20100810
- fix shared-libtds patch to provide properly library names</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 0.82.1-0.2.20100810dev" date="1297684800">- fix again shared-libtds patch to provide increased library version</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 0.82.1-0.3.20110306dev" date="1299672000">- update to the latest stable snapshot 0.82.1.dev.20110306
- make build with shared-libtds conditional
- disable shared-libtds patch by default (seems noone uses it for now)</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 0.91-1" date="1319630400">- Upgrade to 0.91
- Drop shared-libtds support</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 0.91-2" date="1330344000">- Enable Kerberos support (#797276)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.91-3" date="1451908800">- Added missing libgcrypt dependency</changelog>
</package>
<package pkgid="61b91e2ad88bc307c8c94ac809bc163c00d27efea635f2635673f5ff6ab462d0" name="gearmand" arch="src">
  <version epoch="0" ver="1.1.12" rel="18.el7_9"/>
  <changelog author="Petr Machata &lt;pmachata@redhat.com&gt; - 1.1.12-9" date="1422360000">- Rebuild for boost 1.57.0</changelog>
  <changelog author="Adam Jackson &lt;ajax@redhat.com&gt; 1.1.12-10" date="1426680000">- Re-add Fedora conditional dropped in 1.1.12-1, which had the (probably)
  unintended side-effect of reverting Fedora to sysvinit from systemd.</changelog>
  <changelog author="Kalev Lember &lt;kalevlember@gmail.com&gt; - 1.1.12-11" date="1430568000">- Rebuilt for GCC 5 C++11 ABI change</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.12-12" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="David Tardon &lt;dtardon@redhat.com&gt; - 1.1.12-13" date="1437566400">- rebuild for Boost 1.58</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.12-14" date="1438171200">- Rebuilt for https://fedoraproject.org/wiki/Changes/F23Boost159</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.1.12-15" date="1440676800">- Rebuilt for Boost 1.59</changelog>
  <changelog author="Marcin Juszkiewicz &lt;mjuszkiewicz@redhat.com&gt; - 1.1.12-16" date="1453204800">- gperftools is available on wider selection of architectures - rhbz#1256287</changelog>
  <changelog author="Ralf Cors&#xE9;pius &lt;corsepiu@fedoraproject.org&gt; - 1.1.12-17" date="1454414400">- Rebuilt for Boost 1.60
- Append --disable-silent-rules to %configure.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.12-18" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
</package>
<package pkgid="9447ea8ddd09eaac4b7ca0d5c01820a17f9a4b9e5377684dc82014caec45aadc" name="gearmand" arch="src">
  <version epoch="0" ver="1.1.13" rel="1.el7_9"/>
  <changelog author="Adam Jackson &lt;ajax@redhat.com&gt; 1.1.12-10" date="1426680000">- Re-add Fedora conditional dropped in 1.1.12-1, which had the (probably)
  unintended side-effect of reverting Fedora to sysvinit from systemd.</changelog>
  <changelog author="Kalev Lember &lt;kalevlember@gmail.com&gt; - 1.1.12-11" date="1430568000">- Rebuilt for GCC 5 C++11 ABI change</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.12-12" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="David Tardon &lt;dtardon@redhat.com&gt; - 1.1.12-13" date="1437566400">- rebuild for Boost 1.58</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.12-14" date="1438171200">- Rebuilt for https://fedoraproject.org/wiki/Changes/F23Boost159</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.1.12-15" date="1440676800">- Rebuilt for Boost 1.59</changelog>
  <changelog author="Marcin Juszkiewicz &lt;mjuszkiewicz@redhat.com&gt; - 1.1.12-16" date="1453204800">- gperftools is available on wider selection of architectures - rhbz#1256287</changelog>
  <changelog author="Ralf Cors&#xE9;pius &lt;corsepiu@fedoraproject.org&gt; - 1.1.12-17" date="1454414400">- Rebuilt for Boost 1.60
- Append --disable-silent-rules to %configure.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.12-18" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.13-1" date="1600344000">- Update to release 1.1.13</changelog>
</package>
<package pkgid="2becd73466152bf49a8859de9efed568f1e954f144fd56af178641999d6c9d4f" name="gearmand" arch="src">
  <version epoch="0" ver="1.1.14" rel="1.el7_9"/>
  <changelog author="Kalev Lember &lt;kalevlember@gmail.com&gt; - 1.1.12-11" date="1430568000">- Rebuilt for GCC 5 C++11 ABI change</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.12-12" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="David Tardon &lt;dtardon@redhat.com&gt; - 1.1.12-13" date="1437566400">- rebuild for Boost 1.58</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.12-14" date="1438171200">- Rebuilt for https://fedoraproject.org/wiki/Changes/F23Boost159</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.1.12-15" date="1440676800">- Rebuilt for Boost 1.59</changelog>
  <changelog author="Marcin Juszkiewicz &lt;mjuszkiewicz@redhat.com&gt; - 1.1.12-16" date="1453204800">- gperftools is available on wider selection of architectures - rhbz#1256287</changelog>
  <changelog author="Ralf Cors&#xE9;pius &lt;corsepiu@fedoraproject.org&gt; - 1.1.12-17" date="1454414400">- Rebuilt for Boost 1.60
- Append --disable-silent-rules to %configure.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.12-18" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.13-1" date="1600344000">- Update to release 1.1.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.14-1" date="1600344001">- Update to release 1.1.14</changelog>
</package>
<package pkgid="84ab58d7bd8dee121bac1ee4c20a6fb217205c8101422e777eafa34ef339a07f" name="gearmand" arch="src">
  <version epoch="0" ver="1.1.15" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.12-12" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="David Tardon &lt;dtardon@redhat.com&gt; - 1.1.12-13" date="1437566400">- rebuild for Boost 1.58</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.12-14" date="1438171200">- Rebuilt for https://fedoraproject.org/wiki/Changes/F23Boost159</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.1.12-15" date="1440676800">- Rebuilt for Boost 1.59</changelog>
  <changelog author="Marcin Juszkiewicz &lt;mjuszkiewicz@redhat.com&gt; - 1.1.12-16" date="1453204800">- gperftools is available on wider selection of architectures - rhbz#1256287</changelog>
  <changelog author="Ralf Cors&#xE9;pius &lt;corsepiu@fedoraproject.org&gt; - 1.1.12-17" date="1454414400">- Rebuilt for Boost 1.60
- Append --disable-silent-rules to %configure.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.12-18" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.13-1" date="1600344000">- Update to release 1.1.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.14-1" date="1600344001">- Update to release 1.1.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1600344002">- Update to release 1.1.15</changelog>
</package>
<package pkgid="5087ff196811cb399dc2287d783b93c483aa4f1fb79e59dfd52acac2ed847677" name="gearmand" arch="src">
  <version epoch="0" ver="1.1.16" rel="1.el7_9"/>
  <changelog author="David Tardon &lt;dtardon@redhat.com&gt; - 1.1.12-13" date="1437566400">- rebuild for Boost 1.58</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.12-14" date="1438171200">- Rebuilt for https://fedoraproject.org/wiki/Changes/F23Boost159</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.1.12-15" date="1440676800">- Rebuilt for Boost 1.59</changelog>
  <changelog author="Marcin Juszkiewicz &lt;mjuszkiewicz@redhat.com&gt; - 1.1.12-16" date="1453204800">- gperftools is available on wider selection of architectures - rhbz#1256287</changelog>
  <changelog author="Ralf Cors&#xE9;pius &lt;corsepiu@fedoraproject.org&gt; - 1.1.12-17" date="1454414400">- Rebuilt for Boost 1.60
- Append --disable-silent-rules to %configure.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.12-18" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.13-1" date="1600344000">- Update to release 1.1.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.14-1" date="1600344001">- Update to release 1.1.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1600344002">- Update to release 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1600344003">- Update to release 1.1.16</changelog>
</package>
<package pkgid="cc3b86c849752d1acbf6bcc0e2b45808308135f90854d2b4d30e2b5cf4971748" name="gearmand" arch="src">
  <version epoch="0" ver="1.1.17" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.12-14" date="1438171200">- Rebuilt for https://fedoraproject.org/wiki/Changes/F23Boost159</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.1.12-15" date="1440676800">- Rebuilt for Boost 1.59</changelog>
  <changelog author="Marcin Juszkiewicz &lt;mjuszkiewicz@redhat.com&gt; - 1.1.12-16" date="1453204800">- gperftools is available on wider selection of architectures - rhbz#1256287</changelog>
  <changelog author="Ralf Cors&#xE9;pius &lt;corsepiu@fedoraproject.org&gt; - 1.1.12-17" date="1454414400">- Rebuilt for Boost 1.60
- Append --disable-silent-rules to %configure.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.12-18" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.13-1" date="1600344000">- Update to release 1.1.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.14-1" date="1600344001">- Update to release 1.1.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1600344002">- Update to release 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1600344003">- Update to release 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1600344004">- Update to release 1.1.17</changelog>
</package>
<package pkgid="69f8d0f6b233f59169fbd4f43788eb0f0c00b5ba401c62d60464ed51118c2f34" name="gearmand" arch="src">
  <version epoch="0" ver="1.1.18" rel="1.el7_9"/>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.1.12-15" date="1440676800">- Rebuilt for Boost 1.59</changelog>
  <changelog author="Marcin Juszkiewicz &lt;mjuszkiewicz@redhat.com&gt; - 1.1.12-16" date="1453204800">- gperftools is available on wider selection of architectures - rhbz#1256287</changelog>
  <changelog author="Ralf Cors&#xE9;pius &lt;corsepiu@fedoraproject.org&gt; - 1.1.12-17" date="1454414400">- Rebuilt for Boost 1.60
- Append --disable-silent-rules to %configure.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.12-18" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.13-1" date="1600344000">- Update to release 1.1.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.14-1" date="1600344001">- Update to release 1.1.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1600344002">- Update to release 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1600344003">- Update to release 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1600344004">- Update to release 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.18-1" date="1600344005">- Update to release 1.1.18</changelog>
</package>
<package pkgid="90d25faf8b86454108eb19a92c51cd3780785a63289ca2f789dcc29aa958a392" name="gearmand" arch="src">
  <version epoch="0" ver="1.1.19.1" rel="1.el7_9"/>
  <changelog author="Ralf Cors&#xE9;pius &lt;corsepiu@fedoraproject.org&gt; - 1.1.12-17" date="1454414400">- Rebuilt for Boost 1.60
- Append --disable-silent-rules to %configure.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.12-18" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.13-1" date="1600344000">- Update to release 1.1.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.14-1" date="1600344001">- Update to release 1.1.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1600344002">- Update to release 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1600344003">- Update to release 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1600344004">- Update to release 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.18-1" date="1600344005">- Update to release 1.1.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1600344006">- Update to release 1.1.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19.1-1" date="1600344007">- Update to release 1.1.19.1</changelog>
</package>
<package pkgid="eeef12cb724499d05fcaa5ffd10d6308504a7301a493484c64de170f92eda9d2" name="gearmand" arch="src">
  <version epoch="0" ver="1.1.20" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1.12-18" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.13-1" date="1600344000">- Update to release 1.1.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.14-1" date="1600344001">- Update to release 1.1.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1600344002">- Update to release 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1600344003">- Update to release 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1600344004">- Update to release 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.18-1" date="1600344005">- Update to release 1.1.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1600344006">- Update to release 1.1.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19.1-1" date="1600344007">- Update to release 1.1.19.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1668859200">- Update to release 1.1.20</changelog>
</package>
<package pkgid="7e0d54eeaff592188a4371b16e03d5175f7fb5d41399dcbc000284875ea3ce6f" name="gearmand" arch="src">
  <version epoch="0" ver="1.1.21" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.13-1" date="1600344000">- Update to release 1.1.13</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.14-1" date="1600344001">- Update to release 1.1.14</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.15-1" date="1600344002">- Update to release 1.1.15</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.16-1" date="1600344003">- Update to release 1.1.16</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.17-1" date="1600344004">- Update to release 1.1.17</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.18-1" date="1600344005">- Update to release 1.1.18</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19-1" date="1600344006">- Update to release 1.1.19</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.19.1-1" date="1600344007">- Update to release 1.1.19.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.20-1" date="1668859200">- Update to release 1.1.20</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.21-1" date="1699358400">- Update to release 1.1.21</changelog>
</package>
<package pkgid="a567b8810f4b4889d0f11c959a92baa072c22de4edca27389b64dcc8da953f69" name="gengetopt" arch="src">
  <version epoch="0" ver="2.23" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.22.6-7" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.22.6-8" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.22.6-9" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.22.6-10" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.22.6-11" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.22.6-12" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.22.6-13" date="1548936000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Tim Landscheidt &lt;tim@tim-landscheidt.de&gt; - 2.22.6-14.el7_9" date="1551960000">- Remove obsolete requirements for %post/%preun scriptlets</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.22.6-15" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Mosaab Alzoubi &lt;moceap[At]hotmail[Dot]com&gt; - 2.23 -1" date="1568030400">- Update to 2.23</changelog>
</package>
<package pkgid="304cf7a5b005555e413059ae3a76e36a428a0616e42474559683a4e54ee0b55d" name="glib" arch="src">
  <version epoch="1" ver="1.2.10" rel="41.el7_9"/>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; 1:1.2.10-32" date="1239969600">- remove redundant linkage of libgmodule to libgthread
- cosmetic spec changes</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; 1:1.2.10-33" date="1248436800">- rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; 1:1.2.10-34" date="1297166400">- rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; 1:1.2.10-35" date="1319630400">- rebuilt for glibc bug#747377</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; 1:1.2.10-36" date="1325764800">- rebuilt for gcc 4.7</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:1.2.10-37" date="1343390400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:1.2.10-38" date="1360756800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1:1.2.10-39" date="1368187200">- Avoid having to run autotools at build time (#961283)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:1.2.10-40" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1:1.2.10-41" date="1385035200">- Use format strings properly</changelog>
</package>
<package pkgid="afc57906734ecf841ea629f1435df55163f51396390fadeada09c1faec1d3c14" name="gmime" arch="src">
  <version epoch="0" ver="2.6.23" rel="1.el7_9"/>
  <changelog author="Richard Hughes &lt;rhughes@redhat.com&gt; - 2.6.20-1" date="1392811200">- Update to 2.6.20</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.6.20-2" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Kalev Lember &lt;kalevlember@gmail.com&gt; - 2.6.20-3" date="1406030400">- Rebuilt for gobject-introspection 1.41.4</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.6.20-4" date="1408190400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 2.6.20-5" date="1431950400">- Rebuild (mono4)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.6.20-5" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.6.20-6" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Kalev Lember &lt;klember@redhat.com&gt; - 2.6.20-7" date="1474545600">- BR vala instead of obsolete vala-tools subpackage</changelog>
  <changelog author="Kalev Lember &lt;klember@redhat.com&gt; - 2.6.22-1" date="1481976000">- Update to 2.6.22
- Use license macro for COPYING
- Tighten subpackage dependencies</changelog>
  <changelog author="Kalev Lember &lt;klember@redhat.com&gt; - 2.6.23-1" date="1485000000">- Update to 2.6.23</changelog>
</package>
<package pkgid="089972302d39076aa10df006a61beb1dfc28bb91141db77981e4126d139bec26" name="goaccess" arch="src">
  <version epoch="0" ver="0.9.6" rel="1.el7_9"/>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 0.6.1-1" date="1384948800">- Update to 0.6.1</changelog>
  <changelog author="Eduardo Echeverria &lt;echevemaster@gmail.com&gt; - 0.7-1" date="1389787200">- Update to 0.7</changelog>
  <changelog author="Ville Skytt&#xE4; &lt;ville.skytta@iki.fi&gt; - 0.7-2" date="1390046400">- Build with $RPM_OPT_FLAGS, fix -debuginfo.</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 0.7.1-1" date="1392897600">- Update to 0.7.1</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 0.8-1" date="1400673600">- Update to 0.8</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 0.8.1-1" date="1403092800">- Update to 0.8.1</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 0.8.2-1" date="1406030400">- Update to 0.8.2</changelog>
  <changelog author="Eduardo Echeverria &lt;echevemaster@gmail.com&gt; - 0.8.5-1" date="1417780800">- Update to 0.8.5</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 0.9.3-1" date="1440676800">- Update to 0.9.3</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 0.9.6-1" date="1445947200">- Update to 0.9.6</changelog>
</package>
<package pkgid="ea957e28c1356d7e5f1da547fa270d7466c2523ce7c1b3f492f8da156ba46087" name="google-authenticator" arch="src">
  <version epoch="0" ver="1.04" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0-0.gita096a62.1" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0-0.gita096a62.2" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0-0.gita096a62.3" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0-0.gita096a62.4" date="1408190400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0-0.gita096a62.5" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0-0.gita096a62.6" date="1454500800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Marcel Haerry &lt;mh+fedora@scrit.ch&gt; - 1.03-1" date="1486468800">- Bump to latest version</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.03-2" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.03-3" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Marcel Haerry &lt;mh+fedora@scrit.ch&gt; - 1.04-1" date="1502884800">- Update to latest upstream</changelog>
</package>
<package pkgid="4c97cbad8c49503c6d69623f1714b00d1cfd10615b61ee0a6eabcfc613330324" name="gpac" arch="src">
  <version epoch="0" ver="0.5.1" rel="14.20141206svn.el7_9"/>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.5.0-5" date="1376568000">- Rebuilt for FFmpeg 2.0.x</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.5.0-6.20130820svn" date="1377000000">- Update to 20130820</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.5.0-7.20130914svn" date="1379160000">- Update to 20130914</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.5.0-8.20130914svn" date="1383652800">- Rebuilt for x264/FFmpeg</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.5.0-9.20130914svn" date="1395748800">- Rebuilt for ffmpeg-2.2</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.5.0-10.20130914svn" date="1407412800">- Rebuilt for ffmpeg-2.3</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.5.0-11.20140915svn" date="1410782400">- Update to 20140915
- Some clean ups, fix location of html files.</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.5.0-12.20140915svn" date="1411732800">- Rebuilt for FFmpeg 2.4.x</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.5.0-13.20140915svn" date="1413720000">- Rebuilt for FFmpeg 2.4.3</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.5.1-14.20141206svn" date="1417867200">- Update to svn20141206 - last svn rev 5542
- Fix invalid SONAME - rfbz#3365</changelog>
</package>
<package pkgid="8638fcdadc4aa0bd91a2acb77dd01743eabd30466caf8b2cf029e0b85d412a5b" name="gtk+" arch="src">
  <version epoch="1" ver="1.2.10" rel="77.el7_9"/>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; 1:1.2.10-68" date="1239969600">- remove unused shared library dependencies
- use install -p to maintain timestamps where reasonable
- recode docs as UTF-8
- cosmetic spec changes</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; 1:1.2.10-69" date="1248436800">- rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; 1:1.2.10-70" date="1257854400">- don't own dir %{_datadir}/themes/ (owned by filesystem since F-8, #534097)
- make %files lists more specific</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; 1:1.2.10-71" date="1297252800">- rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; 1:1.2.10-72" date="1325764800">- nobody else likes macros for commands</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:1.2.10-73" date="1342699200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:1.2.10-74" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1:1.2.10-75" date="1368187200">- Avoid having to run autotools at build time (#961362)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:1.2.10-76" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 1:1.2.10-77" date="1385035200">- Use format strings properly</changelog>
</package>
<package pkgid="e6db10196aa730d20d457454bb644fe147043e6fef2d213075552ee47036589f" name="hiera" arch="src">
  <version epoch="1" ver="1.3.4" rel="5.el7_9"/>
  <changelog author="Steve Traylen &lt;steve.traylen@cern.ch&gt; - 1.3.2-1" date="1400068800">- New version 1.3.2</changelog>
  <changelog author="Steve Traylen &lt;steve.traylen@cern.ch&gt; - 1.3.2-2" date="1400068801">- Packaging error</changelog>
  <changelog author="Steve Traylen &lt;steve.traylen@cern.ch&gt; - 1.3.3-1" date="1401796800">- New version 1.3.3, Update to latest ruby guidelines.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3.3-2" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Steve Traylen &lt;steve.traylen@cern.ch&gt; - 1.3.4-1" date="1402488000">- New version 1.3.4</changelog>
  <changelog author="Ha&#xEF;kel Gu&#xE9;mar &lt;hguemar@fedoraproject.org&gt; - 1.3.4-2" date="1431950400">- Fix errors with Puppet4 (patch from Lukas Bezdicka)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3.4-3" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Ga&#xEB;l Chamoulaud &lt;gchamoul@redhat.com&gt; - 1.3.4-4" date="1438257600">- Removed 0001-Fix-errors-with-Puppet-4.patch</changelog>
  <changelog author="Steve Traylen &lt;steve.traylen@cern.ch&gt; - 3.0.1-1" date="1446552000">- New version 3.0.1</changelog>
  <changelog author="Steve Traylen &lt;steve.traylen@cern.ch&gt; - 1 1.3.4-1" date="1450440000">- Revert to version 1.3.4 with an epoch of 1. rhbz#1286826</changelog>
</package>
<package pkgid="bfe3f4ba4d99bb201d92f94957062c069376f2811a47eb8727632308f0ea1128" name="hiredis" arch="src">
  <version epoch="0" ver="0.10.1" rel="3.el7_9"/>
  <changelog author="Shakthi Kannan &lt;shakthimaan [AT] fedoraproject dot org&gt; 0.9.2-1" date="1304078400">- First release.</changelog>
  <changelog author="Shakthi Kannan &lt;shakthimaan [AT] fedoraproject dot org&gt; 0.10.0-1.gitdf203bc328" date="1305028800">- Updated to upstream gitdf203bc328.
- Added TODO to the files.
- Updated to use libhiredis.so.0, libhiredis.so.0.10.</changelog>
  <changelog author="Shakthi Kannan &lt;shakthimaan [AT] fedoraproject dot org&gt; 0.10.0-2" date="1305115200">- Updated devel sub-package description.
- Added optimization flags.
- Remove manual installation of shared objects.
- Use upstream .tar.gz sources.</changelog>
  <changelog author="Shakthi Kannan &lt;shakthimaan [AT] fedoraproject dot org&gt; 0.10.0-3" date="1305547200">- Removed INSTALL_LIB from install target as we use INSTALL_LIBRARY_PATH.
- Use 'client library' in Summary.</changelog>
  <changelog author="Shakthi Kannan &lt;shakthimaan [AT] fedoraproject dot org&gt; 0.10.1-1" date="1325246400">- Updated to upstream 0.10.1-28-gd5d8843.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.10.1-2" date="1326456000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Shakthi Kannan &lt;shakthimaan [AT] fedoraproject dot org&gt; 0.10.1-3" date="1327060800">- Removed Requires redis.</changelog>
</package>
<package pkgid="7f062fa4617a81ee6e2659ffcf875c577c418f608616c7994517a5e96e0f81d6" name="hiredis" arch="src">
  <version epoch="0" ver="0.12.1" rel="1.el7_9"/>
  <changelog author="Shakthi Kannan &lt;shakthimaan [AT] fedoraproject dot org&gt; 0.11.0-1" date="1348920000">- Updated to 0.11.0</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.11.0-2" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.11.0-3" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.11.0-4" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.11.0-5" date="1408190400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 0.12.0-1" date="1422014400">- Update to 0.12.0</changelog>
  <changelog author="David Tardon &lt;dtardon@redhat.com&gt; - 0.12.0-2" date="1422360000">- install all headers</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 0.12.0-3" date="1422619200">- Once build on f22</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 0.12.0-4" date="1422619201">- Again build for f22-boost</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 0.12.1-1" date="1422619202">- Update to 0.12.1</changelog>
</package>
<package pkgid="09ab32c0daa89e506130d187043648d628b6f8e25de4b48d2349f37afa28ff4b" name="hping3" arch="src">
  <version epoch="0" ver="0.0.20051105" rel="24.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.0.20051105-15" date="1248436800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.0.20051105-16" date="1297252800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.0.20051105-17" date="1326456000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Paul Wouters &lt;pwouters@redhat.com&gt; - 0.0.20051105-18" date="1342526400">- Fix typo in output (tramitting -&gt; transmitting), rhbz#781325</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.0.20051105-19" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.0.20051105-20" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Jaroslav &#x160;karvada &lt;jskarvad@redhat.com&gt; - 0.0.20051105-21" date="1400673600">- Rebuilt for https://fedoraproject.org/wiki/Changes/f21tcl86</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.0.20051105-22" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.0.20051105-23" date="1408190400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Marcin Juszkiewicz &lt;mjuszkiewicz@redhat.com&gt; - 0.0.20051105-24" date="1410350400">- Handle AArch64</changelog>
</package>
<package pkgid="114410efe66dbf496492f856c0d104cfe3be1b61b5f8d7ca085cb4d3e6fe487c" name="id3lib" arch="src">
  <version epoch="0" ver="3.8.3" rel="32.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.8.3-23" date="1248436800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="Adrian Reber &lt;adrian@lisas.de&gt; - 3.8.3-24" date="1258027200">- Fix "Stack smashing with vbr mp3 files" (bz #533706)
  also see https://bugs.launchpad.net/ubuntu/+source/id3lib3.8.3/+bug/444466</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.8.3-25" date="1297252800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.8.3-26" date="1326456000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.8.3-27" date="1330430400">- Rebuilt for c++ ABI breakage</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.8.3-28" date="1342699200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.8.3-29" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Adrian Reber &lt;adrian@lisas.de&gt; - 3.8.3-30" date="1375876800">- Remove unneeded parts (clean, defattr, buildroot)
- Added man pages from Debian
- Fixed bogus dates</changelog>
  <changelog author="David King &lt;amigadave@amigadave.com&gt; - 3.8.3-31" date="1394107200">- Use autoreconf patch from mingw-id3lib package</changelog>
  <changelog author="David King &lt;amigadave@amigadave.com&gt; - 3.8.3-32" date="1416052800">- Fix typos in man page patch
- Add UTF-16 string lists patch, adapted from Debian
- Add NULL pointer check patch from Debian
- Enable check, using make check</changelog>
</package>
<package pkgid="5a45384e446e77243486c6669c0c16d4eb17c1ad68b6760d5879ff42b24f1dfd" name="iftop" arch="src">
  <version epoch="0" ver="1.0" rel="0.21.pre4.el7_9"/>
  <changelog author="Robert Scheck &lt;robert@fedoraproject.org&gt; 1.0-0.12.pre4" date="1480420800">- Added patch to fix broken MAC address output (#1063298, #1165349)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0-0.13.pre4" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Robert Scheck &lt;robert@fedoraproject.org&gt; 1.0-0.14.pre4" date="1490097600">- Added patch from upstream to fix DNS resolution (#1120254, #1309755)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0-0.15.pre4" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0-0.16.pre4" date="1501675200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0-0.17.pre4" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0-0.18.pre4" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0-0.19.pre4" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.0-0.20.pre4" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Robert Scheck &lt;robert@fedoraproject.org&gt; 1.0-0.21.pre4" date="1564228800">- Added patch from upstream to choose first "running" interface,
  rather than first "up" interface (#1403025)
- Added patch from upstream to support scales beyond 1 Gbps</changelog>
</package>
<package pkgid="5cd0d23fe7c5335dedfc6aae0c8d1d17b26fa650e42aed563efd116afffedd61" name="iksemel" arch="src">
  <version epoch="0" ver="1.4" rel="2.el7_9"/>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.3-1" date="1188043200">- Update to 1.3
- Don't need to BR pkgconfig because gnutls-devel now requires it properly.
- Update license.
- Run "make check" on PPC
- Update patches so that they apply to 1.3
- Update URLs</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.3-2" date="1188043201">- Disable "make check" on PPC again.</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.3-3" date="1202731200">- Rebuild for GCC 4.3</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@fedoraproject.org&gt; - 1.3-4" date="1203422400">- Autorebuild for GCC 4.3</changelog>
  <changelog author="Tomas Mraz &lt;tmraz@redhat.com&gt; - 1.3-5" date="1214395200">- rebuild with new gnutls
- remove texinfo dir file from buildroot</changelog>
  <changelog author="Dennis Gilmore &lt;dennis@ausil.us&gt; - 1.3-6" date="1222862400">- disable "make check" on sparcv9</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3-7" date="1235476800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3-8" date="1248436800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.4-1" date="1256212800">- Update to 1.4
- Apply patch from upstream so that gnutls autoconf works.
- Update gcrypt-sha patch so that it applies.</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.4-2" date="1269432000">- Add patch from Quentin Armitage to fix alignment errors</changelog>
</package>
<package pkgid="d0c7fb01cf64e3fdc73531b1cbf000ca8e4df66cbccc9057bd611309c1149460" name="ilbc" arch="src">
  <version epoch="0" ver="1.1.1" rel="4.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 0-0.1" date="1319112000">- Initial package</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.1.1-1" date="1336564800">- Ver. 1.1.1</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.1.1-2" date="1345032000">- Add compat symlinks for old apps</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.1.1-3" date="1347451200">- Added licensing info</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.1-4" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
</package>
<package pkgid="cb952f8c66f1eed10b8f083dcb6b00b8b95cc21a6382bc6d3daa34c7eeedf21c" name="jack-audio-connection-kit" arch="src">
  <version epoch="0" ver="0.118.0" rel="1.el7_9"/>
  <changelog author="Andy Shevchenko &lt;andy@smile.org.ua&gt; 0.109.2-3" date="1217246400">- add a new requirement to be ensure we have /etc/security for postinstall
  script (#359291, #456830)
- provide a pulseaudio start script from README.Fedora
- append values for pulse-rt group to the limits.conf
- update README.Fedora regarding to the recent changes</changelog>
  <changelog author="Andy Shevchenko &lt;andy@smile.org.ua&gt; - 0.116.1-1" date="1229256000">- update to last official release
- update URL tag
- update file list accordingly</changelog>
  <changelog author="Andy Shevchenko &lt;andy@smile.org.ua&gt; - 0.116.1-2" date="1230552000">- fix multiarch conflict again (#477718, #341621)</changelog>
  <changelog author="Andy Shevchenko &lt;andy@smile.org.ua&gt; - 0.116.1-3" date="1231070400">- avoid creation of the LaTeX documentation (temporary fix for #477402)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.116.1-4" date="1235563200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild</changelog>
  <changelog author="Andy Shevchenko &lt;andy@smile.org.ua&gt; - 0.116.1-5" date="1245585600">- create file under /etc/security/limits.d instead of limits.conf hack (#506583)
- rename jack-audio-connection-kit.pa to jack.pa in the documentation part</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.116.1-6" date="1248436800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="Dennis Gilmore &lt;dennis@ausil.us&gt; - 0.116.1-7" date="1256644800">- dont build libfreebob support on s390 arches</changelog>
  <changelog author="Tom &quot;spot&quot; Callaway &lt;tcallawa@redhat.com&gt; - 0.116.2-8" date="1257336000">- update to 0.116.2
- make sure we cleanup threads that we open, fixes segfaults (thanks to Ray Strode)</changelog>
  <changelog author="Andy Shevchenko &lt;andy@smile.org.ua&gt; - 0.118.0-1" date="1258804800">- update to 0.118.0 (should fix #533419)
- remove upstreamed patch
- append new binaries to -example-clients subpackage</changelog>
</package>
<package pkgid="fa2a88da160aac0fba586b5b5142cd30c6d1ee822c7c78035a82d4f35419c13c" name="janus-gateway" arch="src">
  <version epoch="0" ver="0.10.3" rel="1.el7_9"/>
  <changelog author="Giacomo Sanchietti &lt;giacomo.sanchietti@nethesis.it&gt; - 0.4.3-2" date="1537876800">- Update to upstream commit ef8477e6081c4015e244fbce37d9930e73b83412</changelog>
  <changelog author="Alessandro Polidori &lt;alessandro.polidori@nethesis.it&gt; - 0.5.0-1" date="1542801600">- Update janus-gateway to 0.5.0 - NethServer/dev#5648</changelog>
  <changelog author="Alessandro Polidori &lt;alessandro.polidori@nethesis.it&gt; - 0.6.0-1" date="1547553600">- Update janus-gateway to 0.6.0 - NethServer/dev#5555</changelog>
  <changelog author="Alessandro Polidori &lt;alessandro.polidori@nethesis.it&gt; - 0.6.1-1" date="1551355200">- Update janus-gateway to 0.6.1 - NethServer/dev#5723</changelog>
  <changelog author="Alessandro Polidori &lt;alessandro.polidori@nethesis.it&gt; - 0.6.2-1" date="1551960000">- Update janus-gateway to 0.6.2 - NethServer/dev#5728</changelog>
  <changelog author="Alessandro Polidori &lt;alessandro.polidori@nethesis.it&gt; - 0.6.3-1" date="1553169600">- Update janus-gateway to 0.6.3 - NethServer/dev#5735</changelog>
  <changelog author="Alessandro Polidori &lt;alessandro.polidori@nethesis.it&gt; - 0.10.1.1-1" date="1592308800">- Upgrade janus-gateway to commit 085ed39 - NethServer/dev#6195</changelog>
  <changelog author="Alessandro Polidori &lt;alessandro.polidori@nethesis.it&gt; - 0.10.2.1-1" date="1592827200">- Upgrade janus-gateway to commit a46344d - NethServer/dev#6195</changelog>
  <changelog author="Davide Principi &lt;davide.principi@nethesis.it&gt; - 0.10.2-2" date="1594728000">- Janus-Gateway: upgrade to 922b392 - NethServer/dev#6195
- Upgrade janus lib to 922b392 - nethesis/dev#5824
- Development builds for Janus and sofia-sip - nethesis/dev#5836</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.10.3-1" date="1596715200">- Upgrade janus-gateway to version 0.10.3
- Changed all spec file to RHEL/CentOS standard paths
- Added devel subpackage</changelog>
</package>
<package pkgid="39591beece81b399c2f79743924374085c294b4981ed0457b55028e7690779ba" name="jitsi" arch="src">
  <version epoch="0" ver="2.11.5633" rel="5.el7_9"/>
  <changelog author="wally &lt;wally&gt; 2.10.5550-5.mga7" date="1540296000">+ Revision: 1324696
- remove requires excludes
- enable debug pkgs
- build on all arches again
- add patch to fix native modules build from sources
- add patches from upstream for native module building</changelog>
  <changelog author="daviddavid &lt;daviddavid&gt; 2.10.5550-6.mga7" date="1540382400">+ Revision: 1324818
- fix build on armv7hl</changelog>
  <changelog author="daviddavid &lt;daviddavid&gt; 2.10.5550-7.mga8" date="1566302400">+ Revision: 1430618
- add patch to fix harfbuzz headers path</changelog>
  <changelog author="umeabot &lt;umeabot&gt; 2.10.5550-8.mga8" date="1581768000">+ Revision: 1528457
- Mageia 8 Mass Rebuild</changelog>
  <changelog author="David Va &lt;davidva AT tuta DOT io&gt; 2.10.5550-9" date="1585310400">- Upstream
- Modernized</changelog>
  <changelog author="David Va &lt;davidva AT tuta DOT io&gt; 2.11.5633-1" date="1585656000">- Updated to 2.11.5633</changelog>
  <changelog author="David Va &lt;davidva AT tuta DOT io&gt; 2.11.5633-2" date="1591185600">- Added requires ffmpeg3-libs</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 2.11.5633-3" date="1626523200">- Fix build on F34+</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 2.11.5633-4" date="1628683200">- fix native libraries and  osgi.wiring.package;
  (&amp;(osgi.wiring.package=com.sun.jna)(version&gt;=5.5.0)(\!(version&gt;=6.0.0)))]]
  dropping the last commit</changelog>
  <changelog author="David Va &lt;davidva AT tuta DOT io&gt; 2.11.5633-5" date="1628769600">- Launcher improvements and splash activation</changelog>
</package>
<package pkgid="935481e26fff10749ce820349e94454ac460724c09e867fb19f8d9cca7d5e062" name="jq" arch="src">
  <version epoch="0" ver="1.6" rel="2.el7_9"/>
  <changelog author="Flavio Percoco &lt;flavio@redhat.com&gt; - 1.3-1" date="1382097600">- Initial package release.</changelog>
  <changelog author="Flavio Percoco &lt;flavio@redhat.com&gt; - 1.3-2" date="1382616000">- Added check, manpage</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3-3" date="1402228800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3-4" date="1408190400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3-5" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Ha&#xEF;kel Gu&#xE9;mar &lt;hguemar@fedoraproject.org&gt; - 1.5-1" date="1440504000">- Upstream 1.5.0</changelog>
  <changelog author="Ha&#xEF;kel Gu&#xE9;mar &lt;hguemar@fedoraproject.org&gt; - 1.6-1" date="1578484800">- Upstream 1.6 release
- Sync with Fedora spec</changelog>
  <changelog author="Troy Dawson&lt;tdawson@redhat.com&gt; - 1.6-2" date="1590580800">- Rebuilt for updated oniguruma (#1836692)</changelog>
</package>
<package pkgid="116f32714ced75755707f768b1dbb2675892edc3a32eaef372390076f3e99a85" name="jsoncpp" arch="src">
  <version epoch="0" ver="0.10.5" rel="2.el7_9"/>
  <changelog author="S&#xE9;bastien Willmann &lt;sebastien.willmann@gmail.com&gt; - 0.6.0-0.2.rc2" date="1354449600">- Changed license field to Public Domain or MIT</changelog>
  <changelog author="Sebastien Willmann &lt;sebastien.willmann@gmail.com&gt; - 0.6.0-0.3.rc2" date="1355313600">- Removed static package
- Preserving timestamp on installed files
- Added guard grep to the sed expression
- Removed duplicated doc files
- Removed dependency on pkgconfig
- Changed base package group</changelog>
  <changelog author="Sebastien Willmann &lt;sebastien.willmann@gmail.com&gt; - 0.6.0-0.4.rc2" date="1356091200">- Removed doc subpackage
- Added .pc file
- Fixed shared lib</changelog>
  <changelog author="Sebastien Willmann &lt;sebastien.willmann@gmail.com&gt; - 0.6.0-0.5.rc2" date="1356177600">- Added libjsoncpp.so.0
- Moved the shared lib build to the correct section</changelog>
  <changelog author="Sebastien Willmann &lt;sebastien.willmann@gmail.com&gt; - 0.6.0-0.6.rc2" date="1358596800">- Install the corrected library</changelog>
  <changelog author="Sebastien Willmann &lt;sebastien.willmann@gmail.com&gt; - 0.6.0-0.7.rc2" date="1358683200">- Added graphviz as a BuildRequire</changelog>
  <changelog author="Sebastien Willmann &lt;sebastien.willmann@gmail.com&gt; - 0.6.0-0.8.rc2" date="1360929600">- Added documentation sub-package</changelog>
  <changelog author="S&#xE9;bastien Willmann &lt;sebastien.willmann@gmail.com&gt; - 0.6.0-0.9.rc2" date="1363348800">- Changed Summary
- Added %doc files to the doc package
- Added python as an explicit BuildRequires</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;fedora@besser82.io&gt; - 0.10.5-1" date="1458907200">- Update to version 0.10.5
- Use %license and %doc properly
- Add generated CMake-target
- Move %check after %install
- Remove Group-tag, needed for el &lt;= 5, only
- Drop Patch0, not needed anymore
- Disabled Werror
- Add disttag
- Use cmake instead of scons</changelog>
  <changelog author="S&#xE9;bastien Willmann &lt;sebastien.willmann@gmail.com&gt; - 0.10.5-2" date="1461412800">- Fix file conflict with json-c</changelog>
</package>
<package pkgid="1996c53af8e34e3d790da8910f44ce1c8ee3be4efd7e867323dbe972ad88c636" name="julius" arch="src">
  <version epoch="0" ver="4.5" rel="4.el7_9"/>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 4.4.2.1-1" date="1510056000">- update to 4.4.2.1
- rename jcontrol to julius-jcontrol to avoid conflict with oracle java</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 4.4.2.1-2" date="1510142400">- fix cpuid.h conditionalization to be more complete</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.4.2.1-3" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.4.2.1-4" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.4.2.1-5" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 4.4.2.1-6" date="1550404800">- Rebuild for readline 8.0</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 4.5-1" date="1559217600">- update to 4.5</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.5-2" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.5-3" date="1580299200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.5-4" date="1595937600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_33_Mass_Rebuild</changelog>
</package>
<package pkgid="e2e2d39007eb0222d172efa318a290ad82d33f95bdce1d745e4ddd70a09c1c57" name="kronosnet" arch="src">
  <version epoch="0" ver="1.18" rel="1.el7_9"/>
  <changelog author="Christine Caulfield &lt;ccaulfie@redhat.com&gt; - 1.9-1" date="1557835200">Rebase to knet 1.9
  Resolves: rhbz#1688880</changelog>
  <changelog author="Christine Caulfield &lt;ccaulfie@redhat.com&gt; - 1.9-2" date="1557835201">add some covscan fixes
  Resolves: rhbz#1688880</changelog>
  <changelog author="Christine Caulfield &lt;ccaulfie@redhat.com&gt; - 1.9-3" date="1558440000">Fix kronosnet-tests dependancies and add workaround for dwz crash
  Resolves: rhbz#1688880</changelog>
  <changelog author="Christine Caulfield &lt;ccaulfie@redhat.com&gt; - 1.10-1" date="1560340800">Rebase to 1.10 for ACL support
  Resolves: rhbz#1688880</changelog>
  <changelog author="Fabio M. Di Nitto &lt;fdinitto@redhat.com&gt; - 1.10-2" date="1571227200">link: Fix memory corruption when too many up/down events are recorded
  Resolves: rhbz#1753517</changelog>
  <changelog author="Fabio M. Di Nitto &lt;fdinitto@redhat.com&gt; - 1.10-3" date="1571227201">PMTUd: Fix MTU calculation when using crypto
  Resolves: rhbz#1736872
  host: Fix defrag buffer reclaim logic that could cause delivery
        of corrupted data
  ResolveS: rhbz#1761711</changelog>
  <changelog author="Fabio M. Di Nitto &lt;fdinitto@redhat.com&gt; - 1.10-4" date="1571313600">Disable fun_pmtud_crypto_test as it can take several hours to run
  Resolves: rhbz#1736872</changelog>
  <changelog author="Christine Caulfield &lt;ccaulfie@redhat.com&gt; - 1.15-1" date="1586952000">Rebase to 1.15
  Resolves: rhbz#1796503</changelog>
  <changelog author="Christine Caulfield &lt;ccaulfie@redhat.com&gt; - 1.16-1" date="1588766400">Rebase to 1.16
  Resolves: rhbz#1796503</changelog>
  <changelog author="Christine Caulfield &lt;ccaulfie@redhat.com&gt; - 1.18-1" date="1600862400">Rebase to 1.18
  Resolves: rhbz#1855301</changelog>
</package>
<package pkgid="53c675b2df6de588a0cb9bb469b74dcc3a28966749821b3783bc26b4e1763b09" name="lame" arch="src">
  <version epoch="0" ver="3.99.5" rel="1.el7_9"/>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 3.96.1-2" date="1099656000">- Remove complex CFLAGS, recent compilers already do a grrreat job.</changelog>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 3.96.1-3" date="1134648000">- Remove the use off _smp_mflags, as build fails on i386 with -j4 (but didn't
  on x86_64, strange).</changelog>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 3.96.1-4" date="1142596800">- Release bump to drop the disttag number in FC5 build.
- Disable/remove static lib.</changelog>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 3.96.1-5" date="1147867200">- Clear not needed executable stack flag bit from the library to make it work
  with selinux, add prelink build requirement (need to remove for old distros).</changelog>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 3.97-1" date="1161000000">- Update to 3.97.</changelog>
  <changelog author="Dag Wieers &lt;dag@wieers.com&gt; - 3.98.2-1" date="1226318400">- Updated to release 3.98.2.</changelog>
  <changelog author="Steve Huff &lt;shuff@vecna.org&gt; - 3.98.4-1" date="1278417600">- Updated to release 3.98.4 (thanks Brandon Ooi!).
- Captured libvorbis and gtk2 dependencies.</changelog>
  <changelog author="Dag Wieers &lt;dag@wieers.com&gt; - 3.99.4-1" date="1327838400">- Updated to release 3.99.4.</changelog>
  <changelog author="Dag Wieers &lt;dag@wieers.com&gt; - 3.99.4-2" date="1328011200">- Added compatibility with ATrpms.</changelog>
  <changelog author="Dag Wieers &lt;dag@wieers.com&gt; - 3.99.5-1" date="1331467200">- Updated to release 3.99.5.</changelog>
</package>
<package pkgid="4b5fd346b3830a2da047874d392f472af8db9b735604f274f2fb005a4ecd257b" name="lcdproc" arch="src">
  <version epoch="0" ver="0.5.7" rel="11.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.5.7-2" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.5.7-3" date="1408276800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Dan Hor&#xE1;k &lt;dan[at]danny.cz&gt; - 0.5.7-4" date="1412424000">- update BR, libusbx-devel is brought in by libftdi-devel</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.5.7-5" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.5.7-6" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.5.7-7" date="1475150400">- Remove ControlGroup from services - rhbz#1324015</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.5.7-8" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.5.7-9" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.5.7-10" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 0.5.7-11" date="1502107200">- Remove unneeded BuildRequires: pth-devel</changelog>
</package>
<package pkgid="a4a6b4c070e0a6a409e534943aa98fd13236a8a789f316c6d378033635b64342" name="leatherman" arch="src">
  <version epoch="0" ver="1.3.0" rel="9.el7_9"/>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 1.2.0-1" date="1504180800">- Upstream update</changelog>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 1.3.0-1" date="1507118400">- Upstream update
- unbundle nowide</changelog>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 1.3.0-2" date="1508414400">- rebuilt</changelog>
  <changelog author="Ha&#xEF;kel Gu&#xE9;mar &lt;hguemar@fedoraproject.org&gt; - 1.3.0-3" date="1509278400">- Merge with James spec + keep compatibility with CentOS SIGs</changelog>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt; - 1.3.0-4" date="1510056000">- Restore catch to devel build (bz#1510392)
- Use make_build macro as per review</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.3.0-5" date="1516708800">- Rebuilt for Boost 1.66</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.3.0-6" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.3.0-7" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.3.0-8" date="1548331200">- Rebuilt for Boost 1.69</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.3.0-9" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
</package>
<package pkgid="92ff108e114d316478946bb5d7829ff5c9a5afa55529a71dc9f18534fbc3d7f3" name="lib3gpp-evs" arch="src">
  <version epoch="0" ver="15.1.0" rel="2.el7_9"/>
</package>
<package pkgid="309a18dfc2fe31fa2b5ea745eb3b3155cdd9885d3170270c051fa50b321bd28a" name="lib3gpp-evs" arch="src">
  <version epoch="0" ver="15.2.2" rel="1.el7_9"/>
</package>
<package pkgid="c61fcc2f4d61825f566274a697cda377505cf735bfa469819295f47018511cba" name="lib3gpp-evs" arch="src">
  <version epoch="0" ver="15.3.0" rel="1.el7_9"/>
</package>
<package pkgid="ee141e53ffee652160cbdc146fd9e2feee3f9ebca54f172a8cd44cf95d235e66" name="lib3gpp-evs" arch="src">
  <version epoch="0" ver="15.4.0" rel="1.el7_9"/>
</package>
<package pkgid="d32f427f5a974c0cfb051f3519abb47ebb2df196076fe1fcd6118a58dbf95881" name="lib3gpp-evs" arch="src">
  <version epoch="0" ver="16.1.0" rel="1.el7_9"/>
</package>
<package pkgid="86e4e742a92492756b5e3a5fd2f72a38f78349de95c43b7cb1392dcc5a063232" name="lib3gpp-evs" arch="src">
  <version epoch="0" ver="16.2.0" rel="1.el7_9"/>
</package>
<package pkgid="d2ce4f2ffeb2ab48edcacf501ffeedbc78bffccf5cd6c98c2fe33c5762129815" name="lib3gpp-evs" arch="src">
  <version epoch="0" ver="16.3.0" rel="1.el7_9"/>
</package>
<package pkgid="e0f50f26a0fc7b12cb6eda957b0802efb755ffe70721e34e57709f3d8be839c4" name="lib3gpp-evs" arch="src">
  <version epoch="0" ver="17.0.0" rel="1.el7_9"/>
</package>
<package pkgid="bf5715865d6d225847c015b82d4fbccb7ca878bfc53af399e0cda7023ffdf8c7" name="lib3gpp-evs" arch="src">
  <version epoch="0" ver="18.0.0" rel="1.el7_9"/>
</package>
<package pkgid="f1bc4b06967a2fa92381e07edc85c0a92e0ae32e7e2293d67440769189063378" name="libass" arch="src">
  <version epoch="0" ver="0.13.4" rel="1.el7_9"/>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 0.10.2-1" date="1395230400">- Update to 0.10.2
- Run make check</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.10.2-2" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.10.2-3" date="1408276800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Martin Sourada &lt;mso@fedoraproject.org&gt; - 0.12.0-1" date="1415275200">- Codebase moved from google code to github.
- Update to 0.12.0</changelog>
  <changelog author="Martin Sourada &lt;mso@fedoraproject.org&gt; - 0.12.1-1" date="1430049600">- Update to 0.12.1</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.12.1-2" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Martin Sourada &lt;mso@fedoraproject.org&gt; - 0.13.1-1" date="1450353600">- Update to 0.13.1</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.13.1-2" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Martin Sourada &lt;mso@fedoraproject.org&gt; - 0.13.3-1" date="1474977600">- Update to 0.13.3</changelog>
  <changelog author="Martin Sourada &lt;mso@fedoraproject.org&gt; - 0.13.4-1" date="1475841600">- Update to 0.13.4
- Fixes CVE-2016-7969, CVE-2016-7970, CVE-2016-7972</changelog>
</package>
<package pkgid="8a85bd6789104037db6e8350c8a1b69fd972de66a89f3f430a485e63d573ff0d" name="libass" arch="src">
  <version epoch="0" ver="0.13.4" rel="6.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.12.1-2" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Martin Sourada &lt;mso@fedoraproject.org&gt; - 0.13.1-1" date="1450353600">- Update to 0.13.1</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.13.1-2" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Martin Sourada &lt;mso@fedoraproject.org&gt; - 0.13.3-1" date="1474977600">- Update to 0.13.3</changelog>
  <changelog author="Martin Sourada &lt;mso@fedoraproject.org&gt; - 0.13.4-1" date="1475841600">- Update to 0.13.4
- Fixes CVE-2016-7969, CVE-2016-7970, CVE-2016-7972</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.13.4-2" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.13.4-3" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.13.4-4" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.13.4-5" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Simone Caronni &lt;negativo17@gmail.com&gt; - 0.13.4-6" date="1528977600">- Add patch that fixes glyph bounding box size; this bug causes frames to be
  dropped on 2160p+ displays, as this check is too small for resolutions above
  1440p.
- Revamp SPEC file.</changelog>
</package>
<package pkgid="986ec334b12be498051a577ca9dad84cddc13fac5907d9510732c84981e2d973" name="libbson" arch="src">
  <version epoch="0" ver="1.3.5" rel="7.el7_9"/>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.3.1-1" date="1453464000">- Packaged</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.3.3-1" date="1455537600">- 1.3.3 bump</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.3.4-1" date="1458043200">- 1.3.4 bump</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.3.5-1" date="1459425600">- 1.3.5 bump</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.3.5-2" date="1472472000">- Fix buffer overrun/read in bson_strndup</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.3.5-3" date="1476360000">- Do not panic on allocating zero bytes</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.3.5-4" date="1505476800">- Fix CVE-2017-14227 (a crash when parsing an empty code string of
  a codewscope type) (bug #1494402)</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.3.5-5" date="1513857600">- Fix validating dbpointer when parsing BSON (CDRIVER-2339)
- Fix handling UTF-8 two-byte NUL character (CDRIVER-2401)</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.3.5-6" date="1537272000">- Fix CVE-2018-16790 (heap-based buffer over-read in
  _bson_iter_next_internal()) (bug #1627925)</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.3.5-7" date="1649332800">- Fix CVE-2022-24795 (an integer overflow leading to heap memory corruption
  when dealing with large inputs) (bug #2072913)</changelog>
</package>
<package pkgid="ccf256f9b7c9360f91c666922db1d7931431f25dd1cc03579c1cf8f54d572ab1" name="libc-client" arch="src">
  <version epoch="0" ver="2007e" rel="11.el7_9"/>
  <changelog author="Nalin Dahyabhai &lt;nalin@redhat.com&gt; 2002e-16" date="1132747200">- rebuild</changelog>
  <changelog author="Nalin Dahyabhai &lt;nalin@redhat.com&gt; 2002e-17" date="1133438400">- account for the /usr/share/ssl -&gt; /etc/pki/tls move in the patch which
  sets the locations at compile-time (more of #165967)</changelog>
  <changelog author="Nalin Dahyabhai &lt;nalin@redhat.com&gt; 2002e-18" date="1133438401">- rebuild</changelog>
  <changelog author="Jesse Keating &lt;jkeating@redhat.com&gt;" date="1134129600">- rebuilt</changelog>
  <changelog author="Jonathan Kamens &lt;jik@kamens.brookline.ma.us&gt; 2004g-1" date="1136462400">- Upstream version 2004g (#176974)
- Remove obsolete doc file "WARNING"
- Remove security patch included in new upstream version
- Custom flock code no longer necessary; included in upstream</changelog>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; 2004g-2" date="1138708800">- bump soname (#179017)</changelog>
  <changelog author="Jesse Keating &lt;jkeating@redhat.com&gt; - 2004g-2.1" date="1139313600">- rebuilt for new gcc4.1 snapshot and glibc changes</changelog>
  <changelog author="Jesse Keating &lt;jkeating@redhat.com&gt; - 2004g-2.2" date="1139572800">- bump again for double-long bug on ppc(64)</changelog>
  <changelog author="Jesse Keating &lt;jkeating@redhat.com&gt; - 2004g-2.2.1" date="1152705600">- rebuild</changelog>
  <changelog author="Joe Orton &lt;jorton@redhat.com&gt; - 2007e-11" date="1273060800">- update to 2007e, merge with Fedora uw-imap spec (#586875)</changelog>
</package>
<package pkgid="5be9095791a46017f2984dac8b87d9bea52b4fa1f38caa33c5907556613f1c52" name="libconfuse" arch="src">
  <version epoch="0" ver="3.2.1" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.7-8" date="1408276800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.7-9" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.7-10" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Jon Ciesla &lt;limburgher@gmail.com&gt; - 3.0-1" date="1465905600">- New upstream URL, latest release.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.0-2" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Gwyn Ciesla &lt;limburgher@gmail.com&gt; - 3.1-1" date="1495713600">- 3.1, BZ 1455367</changelog>
  <changelog author="Gwyn Ciesla &lt;limburgher@gmail.com&gt; - 3.2-1" date="1496664000">- 3.2, BZ 1458525</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.2-2" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.2-3" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Gwyn Ciesla &lt;limburgher@gmail.com&gt; - 3.2.1-1" date="1503057600">- 3.2.1, BZ 1482712</changelog>
</package>
<package pkgid="38d648b786839a10a4baeda68dcb07e7d1afdf94546f36cc41f2199b86759a15" name="libdc1394" arch="src">
  <version epoch="0" ver="2.2.2" rel="3.el7_9"/>
  <changelog author="Tim Niemueller &lt;tim@niemueller.de&gt; - 2.1.4-1" date="1325678400">- Update to latest stable release 2.1.4</changelog>
  <changelog author="Tim Niemueller &lt;tim@niemueller.de&gt; - 2.1.4-2" date="1325678401">- Fix improperly installed tools (bz #593873)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.1.4-3" date="1342699200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Jay Fenlason &lt;fenlason@redhat.com&gt; - 2.2.0-1" date="1355227200">- New upstream version</changelog>
  <changelog author="Jay Fenlason &lt;fenlason@redhat.com&gt; - 2.2.0-2" date="1359460800">- Fix two minor issues detected by rpmdiff: installed binaries not stripped
  and -tools depended on version not version-release</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.2.0-3" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Ville Skytt&#xE4; &lt;ville.skytta@iki.fi&gt; - 2.2.0-4" date="1375876800">- Install docs using %doc (#993839).</changelog>
  <changelog author="Tim Niemueller &lt;tim@niemueller.de&gt; - 2.2.2-1" date="1394712000">- Upgrade to latest stable release 2.2.2</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.2.2-2" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.2.2-3" date="1408276800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
</package>
<package pkgid="9e20a6e7defcf1e4bd2e753729ce022241f35eea5a6599fc90dd723e8503f519" name="libftdi" arch="src">
  <version epoch="0" ver="1.3" rel="9.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2-8" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Jon Ciesla &lt;limburgher@gmail.com&gt; 1.3-1" date="1465905600">- 1.3, libconfuse rebuild.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3-2" date="1468929600">- https://fedoraproject.org/wiki/Changes/Automatic_Provides_for_Python_RPM_Packages</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.3-3" date="1485518400">- Rebuilt for Boost 1.63</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.3-4" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Gwyn Ciesla &lt;limburgher@gmail.com&gt; 1.3-5" date="1495713600">- libconfuse rebuild.</changelog>
  <changelog author="Jonathan Wakely &lt;jwakely@redhat.com&gt; - 1.3-6" date="1499083200">- Rebuilt for Boost 1.64</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.3-7" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Zbigniew J&#x119;drzejewski-Szmek &lt;zbyszek@in.waw.pl&gt; - 1.3-8" date="1503144000">- Python 2 binary package renamed to python2-libftdi
  See https://fedoraproject.org/wiki/FinalizingFedoraSwitchtoPython3</changelog>
  <changelog author="Zbigniew J&#x119;drzejewski-Szmek &lt;zbyszek@in.waw.pl&gt; - 1.3-9" date="1503230400">- Add Provides for the old name without %_isa</changelog>
</package>
<package pkgid="30e66feca1ce44959888e7d14f5fe3254b4aea9da01333cd0641a882c46ec790" name="libinjection" arch="src">
  <version epoch="0" ver="3.10.0" rel="4.el7_9"/>
  <changelog author="Oden Eriksson &lt;oe@nux.se&gt; 3.10.0-0" date="1638792000">- initial RPM package</changelog>
  <changelog author="Oden Eriksson &lt;oe@nux.se&gt; 3.10.0-1" date="1639137600">- fixed according to #2029308</changelog>
  <changelog author="Oden Eriksson &lt;oe@nux.se&gt; 3.10.0-2" date="1639137601">- S1: build it a bit nicer (libool, make)
- provide the test suite as well</changelog>
  <changelog author="Oden Eriksson &lt;oe@nux.se&gt; 3.10.0-3" date="1639310400">- use the correct rpm macros for CFLAGS and LDFLAGS
- P1: use correct version
- P2-P6: add upstream fixes</changelog>
  <changelog author="Neal Gompa &lt;ngompa@fedoraproject.org&gt; - 3.10.0-4" date="1639742400">- Add ldconfig_scriptlets macro for EPEL 7</changelog>
</package>
<package pkgid="43429e7d28f3e12426dc637d9fb013857502fd1bcf97bdc86d085a8b8848ef86" name="libmad" arch="src">
  <version epoch="0" ver="0.15.1b" rel="20.el7_9"/>
  <changelog author="Thorsten Leemhuis &lt;fedora [AT] leemhuis [DOT] info&gt; - 0.15.1b-11" date="1238328000">- rebuild for new F11 features</changelog>
  <changelog author="David Juran &lt;david@juran.se&gt; - 0.15.1b-12" date="1246449600">- fix typo in multiarch patch
- fix ppc64 version (Bz 691)</changelog>
  <changelog author="David Juran &lt;david@juran.se&gt; - 0.15.1b-13" date="1248004800">- ppc asm patch from David Woodhouse (Bz 730)
- rpmlint warnings</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.15.1b-14" date="1327492800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.15.1b-15" date="1335614400">- Don't use multiarch patch when the result is not hardcoded
- Update FPM
- Add patches from lp#534287 and lp#534287</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.15.1b-16" date="1362312000">- Mass rebuilt for Fedora 19 Features</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.15.1b-17" date="1409486400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; -0.15.1b-18" date="1479470400">- Spec file clean-up and review</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.15.1b-19" date="1479902400">- Add patches from gentoo
- Allow debuginfo generation</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.15.1b-20" date="1484827200">- Update Thumb-2 patch - rhbz#1414486</changelog>
</package>
<package pkgid="798703ea5bca8ac9c8492c578844f5e031e4a61e9aa4ce8569b0408944b84fba" name="libmcrypt" arch="src">
  <version epoch="0" ver="2.5.8" rel="13.el7_9"/>
  <changelog author="Tom &quot;spot&quot; Callaway &lt;tcallawa@redhat.com&gt; 2.5.8-4" date="1193659200">- multilib fix (bz 342221)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@fedoraproject.org&gt; - 2.5.8-5" date="1203422400">- Autorebuild for GCC 4.3</changelog>
  <changelog author="Tom &quot;spot&quot; Callaway &lt;tcallawa@redhat.com&gt; - 2.5.8-6" date="1219752000">- apply minor cleanups from upstream</changelog>
  <changelog author="Tom &quot;spot&quot; Callaway &lt;tcallawa@redhat.com&gt; - 2.5.8-7" date="1231156800">- fix multilib conflict (bz 478879)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.5.8-8" date="1235563200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.5.8-9" date="1248436800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.5.8-10" date="1297166400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.5.8-11" date="1326456000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.5.8-12" date="1342699200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.5.8-13" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
</package>
<package pkgid="46a70d8a46e31bbf59e5ef88fb8dc7df9f32626b70fd52a969a28cfeebd1a9fb" name="libmfx" arch="src">
  <version epoch="0" ver="1.23" rel="3.el7_9"/>
  <changelog author="Dominik Mierzejewski &lt;rpm@greysector.net&gt; - 1.16-1.20160317git7adf2e4" date="1468152000">- initial build</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.16-2.20160317git7adf2e4" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Dominik Mierzejewski &lt;rpm@greysector.net&gt; - 1.19-1.20170114gita5ba231" date="1487073600">- update to 1.19 release</changelog>
  <changelog author="Dominik Mierzejewski &lt;rpm@greysector.net&gt; - 1.21-1.20170322gite6fc25c" date="1490529600">- update to 1.21</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.21-2" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.21-3" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Dominik Mierzejewski &lt;rpm@greysector.net&gt; - 1.23-1" date="1508241600">- update to 1.23 (#1471768)
- bump SONAME due to incompatible ABI change in 1.23
  (https://github.com/lu-zero/mfx_dispatch/issues/44)</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;besser82@fedoraproject.org&gt; - 1.23-2" date="1516017600">- Rebuilt for libva.so.2</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.23-3" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
</package>
<package pkgid="b11002eebaadd8d6705fed7f3e1bb66dbed854dc1e733a8fa70f1d8c30fef308" name="libmicrohttpd" arch="src">
  <version epoch="1" ver="0.9.71" rel="1.el7_9"/>
  <changelog author="Bj&#xF6;rn Esser &lt;besser82@fedoraproject.org&gt; - 1:0.9.63-2" date="1556107200">- Remove hardcoded gzip suffix from GNU info pages</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;besser82@fedoraproject.org&gt; - 1:0.9.64-1" date="1556107201">- Update to 1:0.9.64</changelog>
  <changelog author="Martin Gansser &lt;martinkg@fedoraproject.org&gt; - 1:0.9.65-1" date="1562328000">- Update to 1:0.9.65</changelog>
  <changelog author="Martin Gansser &lt;martinkg@fedoraproject.org&gt; - 1:0.9.66-1" date="1566043200">- Update to 1:0.9.66</changelog>
  <changelog author="Martin Gansser &lt;martinkg@fedoraproject.org&gt; - 1:0.9.67-1" date="1571400000">- Update to 1:0.9.67</changelog>
  <changelog author="Martin Gansser &lt;martinkg@fedoraproject.org&gt; - 1:0.9.68-1" date="1572264000">- Update to 1:0.9.68</changelog>
  <changelog author="Martin Gansser &lt;martinkg@fedoraproject.org&gt; - 1:0.9.69-1" date="1576497600">- Update to 1:0.9.69</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1:0.9.69-2" date="1580299200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild</changelog>
  <changelog author="Martin Gansser &lt;martinkg@fedoraproject.org&gt; - 1:0.9.70-1" date="1581249600">- Update to 1:0.9.70</changelog>
  <changelog author="Martin Gansser &lt;martinkg@fedoraproject.org&gt; - 1:0.9.71-1" date="1593432000">- Update to 1:0.9.71</changelog>
</package>
<package pkgid="6afa6070ca66ad218d7675595081ca1023463592a9ef95162ccebb8e819644bc" name="libmikmod" arch="src">
  <version epoch="0" ver="3.2.0" rel="21.el7_9"/>
  <changelog author="Jindrich Novy &lt;jnovy@redhat.com&gt; - 3.2.0-13.beta4" date="1336910400">- update to 3.2.0-beta4</changelog>
  <changelog author="Karsten Hopp &lt;karsten@redhat.com&gt; 3.2.0-14.beta4.1" date="1336910401">- disable altivec (used on PPC only)</changelog>
  <changelog author="Jindrich Novy &lt;jnovy@redhat.com&gt; 3.2.0-1" date="1338984000">- update to stable libmikmod-3.2.0</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; - 3.2.0-15" date="1341403200">- Bump NVR to 15 so it's higher than the betas
- Drop ancient esound, enable alsa
- Modernise spec</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2.0-16" date="1342699200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Hans de Goede &lt;hdegoede@redhat.com&gt; - 3.2.0-17" date="1344168000">- Fix a crash in Player_Start() (rhbz#845782)</changelog>
  <changelog author="Hans de Goede &lt;hdegoede@redhat.com&gt; - 3.2.0-18" date="1347105600">- Fix a crash in align_pointer() (rhbz#855130)</changelog>
  <changelog author="Hans de Goede &lt;hdegoede@redhat.com&gt; - 3.2.0-19" date="1348660800">- Fix an array overflow caused by libmikmod-CVE-2007-6720.patch (rhbz#859050)
- Cleanup the specfile a bit</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2.0-20" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Hans de Goede &lt;hdegoede@redhat.com&gt; - 3.2.0-21" date="1365249600">- Fix stuttering sound and hang on exit for apps using libmikmod's alsa driver
- Remove non standard options configure adds to CFLAGS
- Run autoreconf for aarch64 support (rhbz#925794)</changelog>
</package>
<package pkgid="638f15ef23856b5108ce9b08f4f880366a54fc0ae265b92841c3d52f022b7311" name="libmodplug" arch="src">
  <version epoch="1" ver="0.8.8.5" rel="3.el7_9"/>
  <changelog author="Ville Skytt&#xE4; &lt;ville.skytta@iki.fi&gt; - 1:0.8.8.3-2" date="1304942400">- Don't require /etc/timidity.cfg on EL-6, there is no suitable provider
  package available in it at the moment.</changelog>
  <changelog author="Ville Skytt&#xE4; &lt;ville.skytta@iki.fi&gt; - 1:0.8.8.3-3" date="1305028800">- Drop dependency on /etc/timidity.cfg, it's not worth the 100MB+ it pulls in.</changelog>
  <changelog author="Ville Skytt&#xE4; &lt;ville.skytta@iki.fi&gt; - 1:0.8.8.4-1" date="1312718400">- Update to 0.8.8.4 (security, #728371).</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:0.8.8.4-2" date="1326456000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:0.8.8.4-3" date="1342699200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:0.8.8.4-4" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:0.8.8.4-5" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Ville Skytt&#xE4; &lt;ville.skytta@iki.fi&gt; - 1:0.8.8.5-1" date="1394452800">- Update to 0.8.8.5 (CVE-2013-4233, CVE-2013-4234, #995580).</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:0.8.8.5-2" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:0.8.8.5-3" date="1408276800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
</package>
<package pkgid="d8f0e48377e49866642ede338230e4977a8f078c384f7507a61d2999425c9cc0" name="libmodplug" arch="src">
  <version epoch="1" ver="0.8.9.0" rel="9.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1:0.8.8.5-9" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Ville Skytt&#xE4; &lt;ville.skytta@iki.fi&gt; - 1:0.8.9.0-1" date="1493121600">- Update to 0.8.9.0</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1:0.8.9.0-2" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1:0.8.9.0-3" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1:0.8.9.0-4" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Antonio Trande &lt;sagitter@fedoraproject.org&gt; - 1:0.8.9.0-5" date="1518868800">- Use %ldconfig_scriptlets</changelog>
  <changelog author="Antonio Trande &lt;sagitterATfedoraproject.org&gt; - 1:0.8.9.0-6" date="1519300800">- Add gcc gcc-c++ BR</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1:0.8.9.0-7" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1:0.8.9.0-8" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1:0.8.9.0-9" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
</package>
<package pkgid="5206975aafef7c6bc8b07ad1b5f1e6a6864b914abc23e5657b1032697f256078" name="libmp4v2" arch="src">
  <version epoch="0" ver="2.0.0" rel="2.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.5.0.1-10" date="1248436800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.5.0.1-11" date="1297166400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.5.0.1-12" date="1326456000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.5.0.1-13" date="1342699200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.5.0.1-14" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.5.0.1-15" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.5.0.1-16" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.5.0.1-17" date="1408276800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 2.0.0-1" date="1420804800">Sat Mar 01 2014 Avi Alkalay &lt;avibrazil@gmail.com&gt;
  - included some documentation

  Mon Aug 02 2010 Honore Doktorr &lt;hdfssk@gmail.com&gt;
  - update to upstream 2.0.0

  Mon Aug 02 2010 François Kooman &lt;fkooman@tuxed.net&gt;
  - update to upstream 1.9.1
  - drop redundant patches
  - move README to main package
  - add cli-manuals to main package
  - no longer include the API documentation in devel package
  - move headers to /usr/include/mp4v2/*
  - remove *.la in install phase instead of excluding it while packaging</changelog>
  <changelog author="Rex Dieter &lt;rdieter@fedoraproject.org&gt; - 2.0.0-2" date="1420891200">- track library soname, so bumps aren't a surprise
- -devel: own %_includedir/mp4v2/</changelog>
</package>
<package pkgid="baeecf89078ed42e41b96d417bfe21f59f99c4a9536510a5bc1984b9511f65fd" name="libnice" arch="src">
  <version epoch="0" ver="0.1.17" rel="2.el7_9"/>
  <changelog author="Kamil Dudka &lt;kdudka@redhat.com&gt; - 0.1.16-2" date="1557489600">- reintroduce autoreconf in %prep</changelog>
  <changelog author="Stefan Becker &lt;chemobejk@gmail.com&gt; - 0.1.16-3" date="1557489601">- test-new-dribble got renamed to test-new-trickle</changelog>
  <changelog author="Stefan Becker &lt;chemobejk@gmail.com&gt; - 0.1.16-4" date="1559649600">- add upstream patch to make audio connection more reliable (#1716936)</changelog>
  <changelog author="Stefan Becker &lt;chemobejk@gmail.com&gt; - 0.1.16-5" date="1561723200">- update to 0.1.16-16-gb998547
- ignore interfaces with "br-", "docker", "veth", "virbr" or "vnet" prefix (#1723770)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.16-6" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.16-7" date="1580299200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild</changelog>
  <changelog author="Stefan Becker &lt;chemobejk@gmail.com&gt; - 0.1.16-8" date="1582891200">- add upstream patches for incorrectly ignored interfaces (#1808410)</changelog>
  <changelog author="Kalev Lember &lt;klember@redhat.com&gt; - 0.1.16-9" date="1584014400">- Rebuilt for gupnp 1.2</changelog>
  <changelog author="Stefan Becker &lt;chemobejk@gmail.com&gt; - 0.1.17-1" date="1590235200">- Update to 0.1.17</changelog>
  <changelog author="Stefan Becker &lt;chemobejk@gmail.com&gt; - 0.1.17-2" date="1590235201">- Update to 0.1.17 (#1839335)</changelog>
</package>
<package pkgid="1bedfe4971de8caca0221c927513c4fe48e836424875a1c8316a3d1f5e9943d4" name="libpri" arch="src">
  <version epoch="0" ver="1.4.13" rel="1.el7_9"/>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.4.11.2-1" date="1276171200">- 1.4.11.2:
-
- This release fixes situation where Q.SIG calling name in FACILITY
- message was not reported to the upper layer:
-
- * pri_facility.c: Q.SIG calling name in FACILITY message not reported
-   to the upper layer. Q.SIG can send the CallingName, CalledName, and
-   ConnectedName in stand alone FACILITY messages.  If the CallingName
-   was not sent in the SETUP message, the caller id name was not
-   reported to the upper layer.  (Closes issue #17458. Reported, tested
-   by: jsmith. Patched by rmudgett)
-
- For a full list of changes in the current release, please see the
- ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11.2
-
- 1.4.11.1:
-
- This release fixes a regression in multi component FACILITY messages
- and includes a minor bug fix for BRI spans:
-
- * Multi component FACILITY messages only process the first
-   component. The code was only processing the first ROSE component in
-   the facility message.  (Closes issue #17428. Reported, tested by:
-   patrol-cz. Patched by rmudgett)
-
- * Inband disconnect setting does nothing on BRI spans.  The
-   acceptinbanddisconnect flag is not inherited when creating a new TEI
-   and thus rendering the setting (and its respective equivalent in
-   Asterisk) a no-op on BRI setups.  (Closes issue #15265. Reported,
-   patched, tested by: paravoid)
-
- For a full list of changes in the current release, please see the
- ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11.1</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.4.11.3-1" date="1280750400">- 1.4.11.3:
- This release fixes a regression in the calling number assignment logic:
-
-  * Calling Number assignment logic change in libpri 1.4.11. Restored the old
-    behaviour if there is more than one calling number in the incoming SETUP
-    message.  A network provided number is reported as ANI.
-    (Closes issue #17495. Reported and tested by ibercom. Patched by rmudgett)
-
- For a full list of changes in the current release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.11.3</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.4.12-0.1.beta1" date="1280750401">- 1.4.12-beta1
-
- This beta release contains some fixes and several new features, among them:
-
- 1.) ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support
-
- 2.) ETSI Advice Of Charge (AOC) support
-
- 3.) ETSI Explicit Call Transfer (ECT) support
-
- 4.) ETSI Call Waiting support for ISDN phones
-
- 5.) ETSI Malicious Call ID support
-
- For a full list of changes in the current release candidate, please see
- the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.12-beta1</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.4.12-0.2.beta2" date="1286539200">- The following are some of the issues resolved in this beta release:
-
-   * Fix issue where calling name is not successfully processed on inbound
-     QSIG PRI calls from Mitel PBX.
-     (Closes issue #17619. Reported by: jims8650. Patched by rmudgett)
-
-   * Added missing code specified by Q.921 (Figure B.8 Page 85) when receive
-     RNR in "Timer Recovery" state.
-     (Closes issue #16791. Reported by: alecdavis. Patched by alecdavis)
-
-   * Fixed issue where incoming calls specifying the channel using a slot
-     map could not negotiate a B channel correctly.
-
-   * Add support to receive ECMA-164 2nd edition OID name ROSE messages.
-
-   * Fixed issue where ISDN BRI PTMP TE does not recover from line faults.
-     (Closes issue #17570. Reported by: jcovert. Patched by rmudgett)
-
-   * Q.921 improvements from comparing Q.921 SDL diagrams with implementation.
-
-   * Q.921/Q.931 message debug output improvements.
-
- This beta release contains several new features, among them:
-
- 1.) ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support
- 2.) ETSI Advice Of Charge (AOC) support
- 3.) ETSI Explicit Call Transfer (ECT) support
- 4.) ETSI Call Waiting support for ISDN phones
- 5.) ETSI Malicious Call ID support
-
- For a full list of changes in the current release candidate, please see
- the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.12-beta2</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4.12-0.2.beta2" date="1297166400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.4.12-0.3.beta3" date="1297166401">-
- The following are some of the issues resolved in this beta release:
-
-   * Prevent a CONNECT message from sending a CONNECT ACKNOWLEDGE in the
-     wrong state.
-     (issue #17360. Reported by: shawkris. Patched by rmudgett)
-
-   * Made Q.921 delay events to Q.931 if the event could immediately
-     generate response frames.
-     (closes issue #17360. Reported by: shawkris. Patched by rmudgett)
-
-   * BRI PTMP: Active channels not cleared when the interface goes down.
-     (closes issue #17865. Reported by: wimpy. Patched by rmudgett)
-
-   * Segfault in pri_schedule_del() - ctrl value is invalid.
-     (closes issue #17522)
-     (closes issue #18032. Reported by: schmoozecom. Patched by rmudgett)
-
-   * Crash when receiving an unknown/unsupported message type.
-     (closes issue #17968. Reported by: gelo. Patched by rmudgett)
-
-   * B410P gets incoming call packets on ISDN but Asterisk doesn't see the
-     call.
-     (closes issue #18232. Reported by: lelio. Patched by rmudgett)
-
-   * SABME flood on backup D-channel in NFAS configuration.
-     (closes issue #18255. Reported by: bklang. Patched by rmudgett)
-
-   * Asterisk is getting a "No D-channels available!" warning message every
-     4 seconds.
-     (closes issue #17270. Reported by: jmls. Patched by rmudgett)
-
- This beta release contains several new features, among them:
-
- 1.) ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support
- 2.) ETSI Advice Of Charge (AOC) support
- 3.) ETSI Explicit Call Transfer (ECT) support
- 4.) ETSI Call Waiting support for ISDN phones
- 5.) ETSI Malicious Call ID support
-
- For a full list of changes in the current release candidate, please see
- the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.12-beta3</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.4.12-1" date="1309953600">- The Asterisk Development Team announces the release of libpri version
- 1.4.12.  This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/libpri/
-
- The following are some of the issues resolved in this release:
-
-  * Add call transfer exchange of subaddresses support and fix PTMP call
-    transfer signaling.
-
-  * Invalid PTMP redirecting signaling as TE towards NT.
-
-  * Add Q931_IE_TIME_DATE to CONNECT message when in network mode.
-    (issue #18047 (JIRA PRI-114). Reported by: wuwu. Patched by rmudgett)
-
-  * Swap of master/slave in pri_enslave() incorrect.
-    (issue #18769 (JIRA PRI-120). Reported by: jcollie. Patched by jcollie)
-
-  * Fix I-frame retransmission quirks.
-
-  * Crash if NFAS swaps D channels on a call with an active timer.
-
-  * DMS-100 not receiving caller name anymore.
-    (issue #18822 (JIRA PRI-121). Reported by: cmorford. Patched by rmudgett)
-
-  * B channel lost by incoming call in BRI NT PTMP mode.
-
-  * Implement the mandatory T312 timer for NT PTMP broadcast SETUP calls.
-
- This release contains several new features, among them:
-
- 1.) ETSI and Q.SIG Call Completion Supplementary Service (CCSS) support
- 2.) ETSI Advice Of Charge (AOC) support
- 3.) ETSI Explicit Call Transfer (ECT) support
- 4.) ETSI Call Waiting support for ISDN phones
- 5.) ETSI Malicious Call ID support
- 6.) Add Display IE text handling options.
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/libpri/releases/ChangeLog-1.4.12</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4.12-2" date="1326456000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4.12-3" date="1342699200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Jeffrey Ollie &lt;jeff@ocjtech.us&gt; - 1.4.13-1" date="1349784000">- The Asterisk Development Team has announced the release of libpri 1.4.13.
- This release is available for immediate download at
- http://downloads.asterisk.org/pub/telephony/libpri
-
- The release of libpri 1.4.13 resolves several issues reported by the
- community and would have not been possible without your participation.
- Thank you!
-
- The following are the issues resolved in this release:
-
- * --- Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8,
-       and B410P cards.
-   (Issue AST-598. Reported by Trey Blancher)
-
- * --- Implement handling a multi-channel RESTART request.
-   (Closes issue PRI-93. Reported by Marcin Kowalczyk)
-
- * --- Removed MDL/TEI management configuration warning message.
-   (Closes issue PRI-137. Reported by Bart Coninckx)
-
- * --- Allow passing compiler flags (CFLAGS, LDFLAGS)
-   (Closes issue PRI-144. Reported by Tzafrir Cohen)
-
- For a full list of changes in this release, please see the ChangeLog:
-
- http://downloads.asterisk.org/pub/telephony/libpri/ChangeLog-1.4.13</changelog>
</package>
<package pkgid="ef5eae28e297068130556524196fee7553dadb86d44ee55c5d1ba33beb50875f" name="libresample" arch="src">
  <version epoch="0" ver="0.1.3" rel="12.el7_9"/>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.1.3-3" date="1156766400">- Bump release and rebuild.</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.1.3-4" date="1188302400">- Update license tag.</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.1.3-5" date="1202731200">- Rebuild for GCC 4.3</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.1.3-6" date="1221134400">- Add a patch that switches to cmake for building and build a shared library.</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.1.3-7" date="1224072000">- Relax CMake requirements</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.1.3-8" date="1224158400">- Add patch</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.1.3-9" date="1224158401">- Update patch</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.1.3-10" date="1235563200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild</changelog>
  <changelog author="Milos Jakubicek &lt;xjakub@fi.muni.cz&gt; - 0.1.3-11" date="1240315200">- Fix FTBFS: fixed failing tests in %check</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.1.3-12" date="1248523200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
</package>
<package pkgid="16f862afad76da26086516e5cdb026d4cf6c710543220c3836ead2396f7bff8c" name="libsodium" arch="src">
  <version epoch="0" ver="1.0.18" rel="1.el7_9"/>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 0.6.1-1" date="1405598400">- Update to 0.6.1</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 0.7.0-1" date="1408881600">- Update to 0.7.0</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 1.0.0-1" date="1413633600">- Update to 1.0.0</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 1.0.1-1" date="1416657600">- Update to 1.0.1</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 1.0.2-1" date="1422014400">- Update to 1.0.2</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 1.0.3-1" date="1436788800">- Update to 1.0.3</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 1.0.5-1" date="1445774400">- Update to 1.0.5</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 1.0.16-1" date="1516190400">- update to 1.0.16
- soname bump to 23 (libsodium13 compat package exists)
- manage update from libsodium23 (3rd party repository)
- Add static library subpackage
- fix license management</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 1.0.17-1" date="1546862400">- update to 1.0.17</changelog>
  <changelog author="Remi Collet &lt;remi@remirepo.net&gt; - 1.0.18-1" date="1559563200">- update to 1.0.18</changelog>
</package>
<package pkgid="9d00b5a3add3de74a0dd97efa5d07d13b58263cf2be28eb529cf7d840ebc8275" name="libsrtp" arch="src">
  <version epoch="0" ver="2.3.0" rel="1.el7_9"/>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 1.5.4-3" date="1456920000">- use upstream provided .pc file (bz1313590)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-4" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-5" date="1501070400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-6" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-7" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-8" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 1.5.4-9" date="1532347200">- add BuildRequires: gcc</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-10" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-11" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-1" date="1578398400">- update to 2.3.0</changelog>
</package>
<package pkgid="4c11b927b1d46a58322ece3a1c9cde984b140bba2058091d71ac28738b38b291" name="libsrtp" arch="src">
  <version epoch="0" ver="2.4.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-10" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-11" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-1" date="1578398400">- update to 2.3.0</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-2" date="1580299200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-3" date="1595937600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_33_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-4" date="1602504000">- add -tools subpackage (thanks to Gerd v. Egidy)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-5" date="1611662400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-6" date="1618574400">- Rebuilt for RHEL 9 BETA on Apr 15th 2021. Related: rhbz#1947937</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-7" date="1628510400">- Rebuilt for IMA sigs, glibc 2.34, aarch64 flags
  Related: rhbz#1991688</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.0-1" date="1675771200">- update to 2.4.0</changelog>
</package>
<package pkgid="76504ba85282a0f4253a746c527bb4bf2647b089571e1c40d362578283f927cd" name="libsrtp" arch="src">
  <version epoch="0" ver="2.4.1" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-11" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-1" date="1578398400">- update to 2.3.0</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-2" date="1580299200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-3" date="1595937600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_33_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-4" date="1602504000">- add -tools subpackage (thanks to Gerd v. Egidy)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-5" date="1611662400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-6" date="1618574400">- Rebuilt for RHEL 9 BETA on Apr 15th 2021. Related: rhbz#1947937</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-7" date="1628510400">- Rebuilt for IMA sigs, glibc 2.34, aarch64 flags
  Related: rhbz#1991688</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.0-1" date="1675771200">- update to 2.4.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.1-1" date="1675771201">- update to 2.4.1</changelog>
</package>
<package pkgid="a2c330e4f5518cfb9d06a5aba2708cfd85c5e4722fa4e674c47d02255baea021" name="libsrtp" arch="src">
  <version epoch="0" ver="2.4.2" rel="1.el7_9"/>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-1" date="1578398400">- update to 2.3.0</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-2" date="1580299200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-3" date="1595937600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_33_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-4" date="1602504000">- add -tools subpackage (thanks to Gerd v. Egidy)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-5" date="1611662400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-6" date="1618574400">- Rebuilt for RHEL 9 BETA on Apr 15th 2021. Related: rhbz#1947937</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-7" date="1628510400">- Rebuilt for IMA sigs, glibc 2.34, aarch64 flags
  Related: rhbz#1991688</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.0-1" date="1675771200">- update to 2.4.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.1-1" date="1675771201">- update to 2.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.2-1" date="1675771202">- update to 2.4.2</changelog>
</package>
<package pkgid="c34e121efdda4154fd47e69d6617288b47b863650114e07932300e22654d78ca" name="libsrtp" arch="src">
  <version epoch="0" ver="2.5.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-10" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.5.4-11" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-1" date="1578398400">- update to 2.3.0</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-2" date="1580299200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-3" date="1595937600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_33_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3.0-4" date="1602504000">- add -tools subpackage (thanks to Gerd v. Egidy)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-5" date="1611662400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_34_Mass_Rebuild</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-6" date="1618574400">- Rebuilt for RHEL 9 BETA on Apr 15th 2021. Related: rhbz#1947937</changelog>
  <changelog author="Mohan Boddu &lt;mboddu@redhat.com&gt; - 2.3.0-7" date="1628510400">- Rebuilt for IMA sigs, glibc 2.34, aarch64 flags
  Related: rhbz#1991688</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.5.0-1" date="1675771200">- update to 2.5.0</changelog>
</package>
<package pkgid="1fba92e3693f185eed2995f3049c74a3137f813e450f430106593c43ac25f385" name="libss7" arch="src">
  <version epoch="0" ver="1.0.2" rel="1.el7_9"/>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.0.0-1" date="1217332800">- First version for Fedora</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.0.1-1" date="1217937600">- Update to 1.0.1</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.0.1-2" date="1223380800">- Add LICENSE to docs</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.0.1-3" date="1223380801">- Don't forget to add spec file.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0.1-4" date="1235563200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0.1-5" date="1248523200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 1.0.2-1" date="1248782400">- Update to 1.0.2</changelog>
</package>
<package pkgid="96138df6885f4af7ea198d1185314009a9668243aa7cbde654ae2dcb03498bd2" name="libstemmer" arch="src">
  <version epoch="0" ver="0" rel="2.585svn.el7_9"/>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 0-1.585svn" date="1417521600">- Initial packaging</changelog>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 0-2.585svn" date="1420545600">- Removed undefined-non-weak-symbol warnings</changelog>
</package>
<package pkgid="fa207ffcc35c956bc8ac5c6441ecf777c59f5f6009cefc8a65e9aaade3969daa" name="libuv" arch="src">
  <version epoch="1" ver="1.30.1" rel="1.el7_9"/>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1.23.2-1" date="1539259200">- Update to 1.23.2
- https://github.com/libuv/libuv/blob/v1.23.2/ChangeLog</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1.24.1-1" date="1547812800">- Update to 1.24.1
- https://github.com/libuv/libuv/blob/v1.24.1/ChangeLog</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1.26.0-1" date="1550059200">- Update to 1.26.0
- https://github.com/libuv/libuv/blob/v1.26.0/ChangeLog</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1.27.0-1" date="1552910400">- Update to libuv 1.27.0
- https://github.com/libuv/libuv/blob/v1.27.0/ChangeLog</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1.28.0-1" date="1556020800">- Update to libuv 1.28.0
- https://github.com/libuv/libuv/blob/v1.28.0/ChangeLog</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1.28.0-2" date="1556884800">- Fix regression in uv_fs_poll_stop() (BZ 1703935)</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1.29.0-1" date="1557921600">- Update to 1.29.0
- Drop upstreamed patch</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1.29.1-1" date="1558440000">- Update to 1.29.1
- https://github.com/libuv/libuv/blob/v1.29.1/ChangeLog</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1.30.0-1" date="1561636800">- Update to 1.30.0
- https://github.com/libuv/libuv/blob/v1.30.0/ChangeLog</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1.30.1-1" date="1562068800">- Update to 1.30.1
- https://github.com/libuv/libuv/blob/v1.30.1/ChangeLog</changelog>
</package>
<package pkgid="620dafd767f0b1c7ef217b8135c66ef06d689d2e6d0f090c18b4bd06bc182631" name="libwebsockets" arch="src">
  <version epoch="0" ver="3.0.1" rel="2.el7_9"/>
  <changelog author="Fabian Affolter &lt;mail@fabian-affolter.ch&gt; - 2.3.0-1" date="1501329600">- Update to latest upstream release 2.3.0 (rhbz#1472509)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-2" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fabian Affolter &lt;mail@fabian-affolter.ch&gt; - 2.4.0-1" date="1508500800">- Update to latest upstream release 2.4.0 (rhbz#1504377)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.4.0-2" date="1518004800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fabian Affolter &lt;mail@fabian-affolter.ch&gt; - 2.4.1-1" date="1518782400">- Update to latest upstream release 2.4.1 (rhbz#1504377)</changelog>
  <changelog author="Fabian Affolter &lt;mail@fabian-affolter.ch&gt; - 2.4.2-1" date="1521115200">- Update to latest upstream release 2.4.2 (rhbz#1504377)</changelog>
  <changelog author="Fabian Affolter &lt;mail@fabian-affolter.ch&gt; - 3.0.0-1" date="1525694400">- Update to latest upstream release 3.0.0 (rhbz#1575605)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.0.0-2" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fabian Affolter &lt;mail@fabian-affolter.ch&gt; - 3.0.1-1" date="1545134400">- Update to latest upstream release 3.0.1 (rhbz#1604687)</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 3.0.1-2" date="1546862400">- Add libuv-devel Requires to devel package</changelog>
</package>
<package pkgid="d36b327cec26bfc9246a138d4cb815389d05ad8bcc652401dd5b3f5d27e22de9" name="lua-rds-parser" arch="src">
  <version epoch="0" ver="0.06" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.06-1" date="1709294400">- Initial package file</changelog>
</package>
<package pkgid="bfc35051e033c68b6b26d3af226034fd11aafab4cea2392aa7b837079735b128" name="lua-resty-balancer" arch="src">
  <version epoch="0" ver="0.05" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.05-1" date="1708776000">- Initial package file</changelog>
</package>
<package pkgid="e4494161423f02237823b4f984dce7943f8555b50e166a53b5dad007512b5f02" name="lua-resty-core" arch="src">
  <version epoch="0" ver="0.1.28" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.1.28-1" date="1708776000">- Initial package file</changelog>
</package>
<package pkgid="694e640063726ecd1da37871179590908c97dc2c5bf047eef2460c42f10d4bf5" name="lua-resty-dns" arch="src">
  <version epoch="0" ver="0.23" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.23-1" date="1708776000">- Initial package file</changelog>
</package>
<package pkgid="b5a9fffdd262b40b68e1efebe37de5de05ab91acc0298cb0eb9405b5758243c6" name="lua-resty-lock" arch="src">
  <version epoch="0" ver="0.09" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.09-1" date="1709294400">- Initial package file</changelog>
</package>
<package pkgid="216ec2217745daff6c81432126245f23f4f58ed569e818ad00caa291256dab8a" name="lua-resty-lrucache" arch="src">
  <version epoch="0" ver="0.13" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.13-1" date="1708776000">- Initial package file</changelog>
</package>
<package pkgid="268feef287d814b2ada6091b337027715a217f26b3c8632c0dd2e265e6dcc5bf" name="lua-resty-memcached" arch="src">
  <version epoch="0" ver="0.17" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.17-1" date="1709294400">- Initial package file</changelog>
</package>
<package pkgid="f6fbf2032ddba141a3b12d59d81bacfbf23846564cc61de944aa0181b858bb64" name="lua-resty-memcached-shdict" arch="src">
  <version epoch="0" ver="0.01rc4" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.01rc4-1" date="1708776000">- Initial package file</changelog>
</package>
<package pkgid="3350c51837a8dd315e7a7439d39d4ca97c3986d5c811a805988fc81dae0f924b" name="lua-resty-mysql" arch="src">
  <version epoch="0" ver="0.27" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.27-1" date="1709294400">- Initial package file</changelog>
</package>
<package pkgid="1163edf8286ebe5c59c7add590040394ae373bb2f74218fd7eb00b29bd52ba7d" name="lua-resty-redis" arch="src">
  <version epoch="0" ver="0.30" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.30-1" date="1709294400">- Initial package file</changelog>
</package>
<package pkgid="b750f739a187c2330c849c4848f63d8a57da57da7be9180d2e8cf39d6fe046a8" name="lua-resty-shdict-simple" arch="src">
  <version epoch="0" ver="0.01rc5" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.01rc5-1" date="1708776000">- Initial package file</changelog>
</package>
<package pkgid="65e53a652ab8137e1b93036da69e707737e35d8193fb8afdc75b52664487caa8" name="lua-resty-shell" arch="src">
  <version epoch="0" ver="0.03" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.03-1" date="1709294400">- Initial package file</changelog>
</package>
<package pkgid="4cb075f7ce57c1776ad0ac65a2ea1c4ecfeef81ad9563bc7f4efddef3832b6be" name="lua-resty-signal" arch="src">
  <version epoch="0" ver="0.03" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.03-1" date="1709294400">- Initial package file</changelog>
</package>
<package pkgid="e795b7c11464956561f8c42c4d3ab26c546691ba95e643d342406c49717e74e7" name="lua-resty-string" arch="src">
  <version epoch="0" ver="0.15" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.15-1" date="1708776000">- Initial package file</changelog>
</package>
<package pkgid="1fd61e24df2542609cb83f8f0efce1027e7a3a797f20f1a494ffb7df47ca3a21" name="lua-resty-upload" arch="src">
  <version epoch="0" ver="0.11" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.11-1" date="1709294400">- Initial package file</changelog>
</package>
<package pkgid="f723e58b0aa6c419b80fdefcfab1feec598c83ca9fe684042b29a938b0eca80b" name="lua-resty-upstream-healthcheck" arch="src">
  <version epoch="0" ver="0.08" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.08-1" date="1709294400">- Initial package file</changelog>
</package>
<package pkgid="2f6c529c33a8e4f51e5d23b353711d7bb92c17a9f571de2b2897bafce8f34693" name="lua-resty-websocket" arch="src">
  <version epoch="0" ver="0.11" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.11-1" date="1708776000">- Initial package file</changelog>
</package>
<package pkgid="b8bc617698204061aa30ec19f94177d2b7f3f4a66e7ce790088186f8fb13934f" name="lua-resty-websocket-proxy" arch="src">
  <version epoch="0" ver="0.0.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.0.1-1" date="1708776000">- Initial package file</changelog>
</package>
<package pkgid="5f36a0632b3d2cae32fc673cfb4ad71015280eb68e76e47f4dfced375bdad461" name="luajit" arch="src">
  <version epoch="0" ver="2.0.4" rel="3.el7_9"/>
  <changelog author="Igor Gnatenko &lt;i.gnatenko.brain@gmail.com&gt; - 2.0.2-7" date="1386331200">- Fix executable binary</changelog>
  <changelog author="Igor Gnatenko &lt;i.gnatenko.brain@gmail.com&gt; - 2.0.2-8" date="1386590400">- Fix strip (thanks Ville Skyttä)</changelog>
  <changelog author="Clive Messer &lt;clive.messer@communitysqueeze.org&gt; - 2.0.2-9" date="1387108800">- Apply luajit-path64.patch on x86_64.</changelog>
  <changelog author="Igor Gnatenko &lt;i.gnatenko.brain@gmail.com&gt; - 2.0.3-1" date="1394712000">- 2.0.3 upstream release</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.0.3-2" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;i.gnatenko.brain@gmail.com&gt; - 2.0.3-3" date="1404907200">- rebuild against lua 5.2</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.0.3-4" date="1408276800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;i.gnatenko.brain@gmail.com&gt; - 2.0.4-1" date="1432209600">- 2.0.4</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.0.4-2" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Oliver Haessler &lt;oliver@redhat.com&gt; - 2.0.4-3" date="1438948800">- only build x86_64 on EPEL as luajit has no support for ppc64</changelog>
</package>
<package pkgid="ef1e57041eeeb75c5c1318a34ff2ba34498da6319aaf704b274d7899d788a5c0" name="luajit" arch="src">
  <version epoch="0" ver="2.0.5" rel="1.el7_9"/>
  <changelog author="Igor Gnatenko &lt;i.gnatenko.brain@gmail.com&gt; - 2.0.2-8" date="1386590400">- Fix strip (thanks Ville Skyttä)</changelog>
  <changelog author="Clive Messer &lt;clive.messer@communitysqueeze.org&gt; - 2.0.2-9" date="1387108800">- Apply luajit-path64.patch on x86_64.</changelog>
  <changelog author="Igor Gnatenko &lt;i.gnatenko.brain@gmail.com&gt; - 2.0.3-1" date="1394712000">- 2.0.3 upstream release</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.0.3-2" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;i.gnatenko.brain@gmail.com&gt; - 2.0.3-3" date="1404907200">- rebuild against lua 5.2</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.0.3-4" date="1408276800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;i.gnatenko.brain@gmail.com&gt; - 2.0.4-1" date="1432209600">- 2.0.4</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.0.4-2" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Oliver Haessler &lt;oliver@redhat.com&gt; - 2.0.4-3" date="1438948800">- only build x86_64 on EPEL as luajit has no support for ppc64</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.0.5-1" date="1708689600">- 2.0.5</changelog>
</package>
<package pkgid="ec643b78ea639be60ef5c877d5592e665d8e9a4ae81e74d9243bc1e382eed573" name="luajit" arch="src">
  <version epoch="0" ver="2.1.0" rel="32.2.el7_9"/>
  <changelog author="tchvatal@suse.com" date="1382011200">- Inital package. Used in mpv multimedia player.</changelog>
  <changelog author="tchvatal@suse.com" date="1382011201">- Make the thing acutally build&amp;install.</changelog>
  <changelog author="tchvatal@suse.com" date="1382529600">- Change lib group to libraries only.</changelog>
  <changelog author="tchvatal@suse.com" date="1382529601">- Drop STRIP=true from the args.</changelog>
  <changelog author="tchvatal@suse.com" date="1382529602">- Export CFLAGS, funny localy it used them properly.</changelog>
  <changelog author="simon@simotek.net" date="1397736000">- %make_install replaced with make DESTDIR=%{buildroot} install
  to fix fedora builds</changelog>
  <changelog author="mailaender@opensuse.org" date="1413633600">- update to version 2.0.3
  * Add support for multilib distro builds.
  * Improve ULOAD forwarding for open upvalues.
  * Fix GC steps threshold handling when called by JIT-compiled code.
  * Fix argument checks for math.deg() and math.rad().
  * Fix jit.flush(func|true).
  * Respect jit.off(func) when returning to a function, too.
  * Fix compilation of string.byte(s, nil, n).
  * Fix line number for relocated bytecode after closure fixup
  * Fix frame traversal for backtraces.
  * Fix ABC elimination.
  * Fix handling of redundant PHIs.
  * Fix snapshot restore for exit to function header.
  * Fix type punning alias analysis for constified pointers
  * Fix call unroll checks in the presence of metamethod frames.
  * Fix initial maxslot for down-recursive traces.
  * Prevent BASE register coalescing if parent uses IR_RETF.
  * Don't purge modified function from stack slots in BC_RET.
  * Fix recording of BC_VARG.
  * Don't access dangling reference to reallocated IR.
  * Fix frame depth display for bytecode dump in -jdump.
  * ARM: Fix register allocation when rematerializing FPRs.
  * x64: Fix store to upvalue for lightuserdata values.
  * FFI: Add missing GC steps for callback argument conversions.
  * FFI: Properly unload loaded DLLs.
  * FFI: Fix argument checks for ffi.string().
  * FFI/x64: Fix passing of vector arguments to calls.
  * FFI: Rehash finalizer table after GC cycle, if needed.
  * FFI: Fix cts-&gt;L for cdata unsinking in snapshot restore.</changelog>
</package>
<package pkgid="cb308d1b92610cf7d19ee3387200f009db1c2ffb013d110ed33050a651eefb4c" name="luajit" arch="src">
  <version epoch="0" ver="2.1.20231006" rel="1.el7_9"/>
  <changelog author="Hiroaki Nakamura &lt;hnakamur@gmail.com&gt; - 2.1.20190530-1" date="1561550400">- 2.1.20190530</changelog>
  <changelog author="Hiroaki Nakamura &lt;hnakamur@gmail.com&gt; - 2.1.20190626-1" date="1563796800">- 2.1.20190626</changelog>
  <changelog author="Hiroaki Nakamura &lt;hnakamur@gmail.com&gt; - 2.1.20190912-1" date="1568462400">- 2.1.20190912</changelog>
  <changelog author="Hiroaki Nakamura &lt;hnakamur@gmail.com&gt; - 2.1.20200102-1" date="1579608000">- 2.1.20200102</changelog>
  <changelog author="Hiroaki Nakamura &lt;hnakamur@gmail.com&gt; - 2.1.20201027-1" date="1603886400">- 2.1.20201027</changelog>
  <changelog author="Hiroaki Nakamura &lt;hnakamur@gmail.com&gt; - 2.1.20201229-1" date="1613563200">- 2.1.20201229</changelog>
  <changelog author="Hiroaki Nakamura &lt;hnakamur@gmail.com&gt; - 2.1.20210510-1" date="1625659200">- 2.1.20210510</changelog>
  <changelog author="Hiroaki Nakamura &lt;hnakamur@gmail.com&gt; - 2.1.20211210-1" date="1641384000">- 2.1.20211210</changelog>
  <changelog author="Hiroaki Nakamura &lt;hnakamur@gmail.com&gt; - 2.1.20220411-1" date="1655899200">- 2.1.20220411</changelog>
  <changelog author="Hiroaki Nakamura &lt;hnakamur@gmail.com&gt; - 2.1.20231006-1" date="1698235200">- 2.1.20231006</changelog>
</package>
<package pkgid="af3ec26b0b0f7580fd49c80a11f868bad588c1b5b066ca6f28653de46ddbb034" name="mISDN" arch="src">
  <version epoch="0" ver="1.1.5" rel="4.el7_9"/>
  <changelog author="David Woodhouse &lt;dwmw2@infradead.org&gt; 0-1.cvs20061010" date="1161086400">- Initial import</changelog>
  <changelog author="David Woodhouse &lt;dwmw2@infradead.org&gt; 1.0.3-1" date="1166270400">- Update to 1.0.3</changelog>
  <changelog author="David Woodhouse &lt;dwmw2@infradead.org&gt; 1.1.0-1" date="1172232000">- Update to 1.1.0</changelog>
  <changelog author="David Woodhouse &lt;dwmw2@infradead.org&gt; 1.1.3-1" date="1180526400">- Update to 1.1.3</changelog>
  <changelog author="David Woodhouse &lt;dwmw2@infradead.org&gt; 1.1.5-1" date="1187784000">- Update to 1.1.5</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@fedoraproject.org&gt; - 1.1.5-2" date="1203336000">- Autorebuild for GCC 4.3</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.5-3" date="1235563200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.5-4" date="1248523200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
</package>
<package pkgid="7fb5d0ba652c2dae63c7ca36130b3d4eb7e4a1efb0178673c3021ba88e350a3d" name="miniupnpc" arch="src">
  <version epoch="0" ver="2.0" rel="3.el7_9"/>
  <changelog author="Kalev Lember &lt;kalevlember@gmail.com&gt; - 1.9-2" date="1401278400">- Rebuilt for https://fedoraproject.org/wiki/Changes/Python_3.4</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.9-3" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.9-4" date="1408276800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.9-5" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="pcpa &lt;paulo.cesar.pereira.de.andrade@gmail.com&gt; - 1.9-6" date="1446292800">- Correct buffer overflow in XML parsing (#1270842, #1270182)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.9-7" date="1447156800">- Rebuilt for https://fedoraproject.org/wiki/Changes/python3.5</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.9-8" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Fran&#xE7;ois Kooman &lt;fkooman@tuxed.net&gt; - 2.0-1" date="1465387200">- update to 2.0</changelog>
  <changelog author="Simone Caronni &lt;negativo17@gmail.com&gt; - 2.0-2" date="1643544000">- Clean up SPEC file.
- Add patch to fix CVE-2017-1000494 (backported from 2.1).
- Drop conditions for Python 3 etc, other branches have moved forward.</changelog>
  <changelog author="Simone Caronni &lt;negativo17@gmail.com&gt; - 2.0-3" date="1643544001">- Add patch to fix CVE-2017-8798 (backported from 2.1).</changelog>
</package>
<package pkgid="0b57b87b9310de5940815cc7017f9c08b8b507b5bfea55e7613a70c4e03bcc02" name="mongo-c-driver" arch="src">
  <version epoch="0" ver="1.3.6" rel="1.el7_9"/>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 1.3.0-1" date="1450267200">- Update to 1.3.0
- move tools in devel package</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 1.3.1-1" date="1453377600">- Update to 1.3.1</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 1.3.2-1" date="1454414400">- Update to 1.3.2</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 1.3.3-1" date="1454846400">- Update to 1.3.3</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 1.3.3-2" date="1456747200">- cleanup for review
- move libraries in "libs" sub-package
- add patch to skip online tests
  open https://github.com/mongodb/mongo-c-driver/pull/314
- temporarily disable test suite on arm  (#1303864)
- temporarily disable test suite on i686/F24+ (#1313018)</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 1.3.4-1" date="1458043200">- update to 1.3.4
- drop patch merged upstream</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 1.3.4-2" date="1458388800">- build with MONGOC_NO_AUTOMATIC_GLOBALS</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 1.3.5-1" date="1459425600">- update to 1.3.5
- use --disable-automatic-init-and-cleanup build option
- ignore check for libbson version = libmongoc version</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 1.3.5-2" date="1463400000">- add patch to enforce system crypto policies</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 1.3.6-1" date="1476964800">- update to 1.3.6</changelog>
</package>
<package pkgid="0a506724e524cf57aa66c0b4d721cd69c1df696cd3aa1ae9e89ee0dc5b4aa048" name="mongodb" arch="src">
  <version epoch="0" ver="2.6.12" rel="6.el7_9"/>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 2.6.8-1" date="1425556800">- Updated to 2.6.8</changelog>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 2.6.9-1" date="1427371200">- Fixed CVE-2015-1609 (#1200449)
- Upgrade to 2.6.9</changelog>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 2.6.10-1" date="1434369600">- Fixed systemd service PIDFile setting (#1231269)
- Upgrade to 2.6.10</changelog>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 2.6.11-1" date="1439812800">- Upgrade to 2.6.11</changelog>
  <changelog author="Marek Skalick&#xFD; &lt;mskalick@redhat.com&gt; - 2.6.12-1" date="1470657600">- Upgrade to 2.6.12</changelog>
  <changelog author="Marek Skalick&#xFD; &lt;mskalick@redhat.com&gt; - 2.6.12-2" date="1470657601">- Disable building for ppc64le (v8 does not support ppc64le)</changelog>
  <changelog author="Marek Skalick&#xFD; &lt;mskalick@redhat.com&gt; - 2.6.12-3" date="1474286400">- Security fix for CVE-2016-6494
  (fix world-readable .dbshell history file)</changelog>
  <changelog author="Marek Skalick&#xFD; &lt;mskalick@redhat.com&gt; - 2.6.12-4" date="1483963200">- Remove 0-length files created by MongoDB logRotate after logrotating
- Update excluded architectures</changelog>
  <changelog author="Richard Shaw &lt;hobbes1069@gmail.com&gt; - 2.6.12-5" date="1515499200">- Rebuild for updated yaml-cpp.</changelog>
  <changelog author="Richard Shaw &lt;hobbes1069@gmail.com&gt; - 2.6.12-6" date="1517227200">- Rebuild for reverting yaml-cpp back to 0.5.1.</changelog>
</package>
<package pkgid="ee4f62603cf0c1fc3217b322363758969e4fbfa5332fdb93c582009412a4a3a5" name="mozjs38" arch="src">
  <version epoch="0" ver="38.8.0" rel="2.el7_9"/>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 38.2.1-1" date="1444132800">- Initial mozjs38 spec (based on mozjs31)</changelog>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 38.2.1-2" date="1444737600">- Fixed inaccuracies from package review (paraller build, tests don't fail,...)</changelog>
  <changelog author="Marcin Juszkiewicz &lt;mjuszkiewicz@redhat.com&gt; - 38.2.1-3" date="1445860800">- Handle all Fedora 64-bit architectures.</changelog>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 38.2.1-4" date="1448452800">- Make two functions from CharacterEncoding.cpp visible (needed by MongoDB)</changelog>
  <changelog author="Dan Hor&#xE1;k &lt;dan[at]danny.cz&gt; - 38.2.1-5" date="1448539200">- fix build on s390 (rhbz#1219542)</changelog>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 38.2.1-6" date="1449230400">- Make one function from vm/Runtime.cpp visible (needed by MongoDB 3.2)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 38.2.1-7" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 38.2.1-8" date="1459425600">- Disable two optimizations which caused regressions with new gcc
- Use fedora default -O2 instead of mozjs -O3</changelog>
  <changelog author="Marek Skalick&#xFD; &lt;mskalick@redhat.com&gt; - 38.8.0-1" date="1473681600">- Use firefox sources
- Upgrade to 38.8.0esr</changelog>
  <changelog author="Marek Skalick&#xFD; &lt;mskalick@redhat.com&gt; - 38.8.0-2" date="1474286400">- Fix aarch64 systems with 48-bit VA
  (patch created by Zheng Xu)</changelog>
</package>
<package pkgid="ff1d9394218d5dbca9d10614f68445b8df5477f920aceab6e5a7a39705f65ad1" name="mpg123" arch="src">
  <version epoch="0" ver="1.20.1" rel="1.el7_9"/>
  <changelog author="Dag Wieers &lt;dag@wieers.com&gt; - 1.14.0-1" date="1339156800">- Updated to release 1.14.0.</changelog>
  <changelog author="Dag Wieers &lt;dag@wieers.com&gt; - 1.14.2-1" date="1339156801">- Updated to release 1.14.2.</changelog>
  <changelog author="Dag Wieers &lt;dag@wieers.com&gt; - 1.14.4-1" date="1343822400">- Updated to release 1.14.4.</changelog>
  <changelog author="Tom G. Christensen &lt;tgc@jupiterrise.com&gt; - 1.14.4-2" date="1346414400">- Plugins need libltdl from libtool</changelog>
  <changelog author="Dag Wieers &lt;dag@wieers.com&gt; - 1.15.1-1" date="1363780800">- Updated to release 1.15.1.</changelog>
  <changelog author="Dag Wieers &lt;dag@wieers.com&gt; - 1.15.4-1" date="1380024000">- Updated to release 1.15.4.</changelog>
  <changelog author="Dag Wieers &lt;dag@wieers.com&gt; - 1.16.0-1" date="1382616000">- Updated to release 1.16.0.</changelog>
  <changelog author="Dag Wieers &lt;dag@wieers.com&gt; - 1.19.0-1" date="1395921600">- Updated to release 1.19.0.</changelog>
  <changelog author="Dag Wieers &lt;dag@wieers.com&gt; - 1.20.0-1" date="1401624000">- Updated to release 1.20.0.</changelog>
  <changelog author="Dag Wieers &lt;dag@wieers.com&gt; - 1.20.1-1" date="1404993600">- Updated to release 1.20.1.</changelog>
</package>
<package pkgid="52268c8a1b8f72a6a45f4f3f848a8dfe4b1f0d786f56a8489c726e051f43739c" name="mysql-connector-python" arch="src">
  <version epoch="0" ver="2.1.3" rel="1.el7_9"/>
  <changelog author="Balasubramanian Kandasamy &lt;balasubramanian.kandasamy@oracle.com&gt; - 2.1.3-1" date="1438344000">- New spec file with support for cext, license options and Python 3 support</changelog>
</package>
<package pkgid="5ba867cc5528d08756f37634e525452b41f6c877936503f816608653879d4bac" name="mysql-utilities" arch="src">
  <version epoch="0" ver="1.6.1" rel="1.el7_9"/>
  <changelog author="Balasubramanian Kandasamy &lt;balasubramanian.kandasamy@oracle.com&gt; - 1.3.6-1" date="1388750400">- initial package</changelog>
  <changelog author="Balasubramanian Kandasamy &lt;balasubramanian.kandasamy@oracle.com&gt; - 1.4.2-1" date="1393416000">- Updated for 1.4.2
- Add extra subpackage</changelog>
  <changelog author="Murthy Narkedimilli &lt;murthy.narkedimilli@oracle.com&gt; - 1.5.0" date="1401105600">- Updated the Version to 1.5.0</changelog>
  <changelog author="Chuck Bell &lt;chuck.bell@oracle.com&gt; - 1.5.1" date="1404216000">- Updated the Version to 1.5.1</changelog>
  <changelog author="Chuck Bell &lt;chuck.bell@oracle.com&gt; - 1.5.2" date="1407758400">- Updated the Version to 1.5.2</changelog>
  <changelog author="Nelson Goncalves &lt;nelson.goncalves@oracle.com&gt; - 1.6.0" date="1407758401">- Updated the Version to 1.6.0</changelog>
  <changelog author="Balasubramanian Kandasamy &lt;balasubramanian.kandasamy@oracle.com&gt; - 1.6.0-1" date="1410264000">- Added new utilities mysqlgrants and mysqlbinlogmove</changelog>
  <changelog author="Murthy Narkedimilli &lt;murthy.narkedimilli@oracle.com&gt; - 1.6.1" date="1418817600">- Added new utilities binaries mysqlbinlogpurge, mysqlbinlogrotate and mysqlslavetrx
- Changed the build prefix for SLES platform.
- Added condition to include the mysql_utilities-*.egg-info file</changelog>
</package>
<package pkgid="7e88b7f9d81c1a8cd1adade7fdb8938e74469d07b0a5bf6127704eb33838bbf5" name="nasm" arch="src">
  <version epoch="0" ver="2.13.03" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.12.02-2" date="1486728000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Mikolaj Izdebski &lt;mizdebsk@redhat.com&gt; - 2.13-1" date="1493726400">- Update to upstream version 2.13</changelog>
  <changelog author="Mikolaj Izdebski &lt;mizdebsk@redhat.com&gt; - 2.13.01-1" date="1495454400">- Update to upstream version 2.13.01</changelog>
  <changelog author="Nils Philippsen &lt;nils@redhat.com&gt; - 2.13.01-2" date="1499169600">- don't build documentation during modular build
- fix bogus changelog date</changelog>
  <changelog author="Mikolaj Izdebski &lt;mizdebsk@redhat.com&gt; - 2.13.01-3" date="1501070400">- Fix use-after-free and heap buffer overflow vulnerabilities
- Resolves: CVE-2017-10686, CVE-2017-11111</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.13.01-4" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Michael Simacek &lt;msimacek@redhat.com&gt; - 2.13.02-1" date="1514894400">- Update to upstream version 2.13.02
- Resolves: CVE-2017-17810, CVE-2017-17811, CVE-2017-17812, CVE-2017-17813,
  CVE-2017-17814, CVE-2017-17815, CVE-2017-17816, CVE-2017-17817,
  CVE-2017-17818, CVE-2017-17819, CVE-2017-17820</changelog>
  <changelog author="Michael Simacek &lt;msimacek@redhat.com&gt; - 2.13.02-2" date="1515672000">- Add missing BR ghostscript</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.13.02-3" date="1518091200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Michael Simacek &lt;msimacek@redhat.com&gt; - 2.13.03-1" date="1518091201">- Update to upstream version 2.13.03</changelog>
</package>
<package pkgid="6c84e1a71519d1f85e5dc0e22358bb63eac894516b4237663b839cdba5114787" name="nginx" arch="src">
  <version epoch="0" ver="1.10.0" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.7-1" date="1449835200">- update to 1.9.7 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.8-1" date="1449835201">- update to 1.9.8 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.9-1" date="1449835202">- update to 1.9.9 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.10-1" date="1455537600">- update to 1.9.10 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.11-1" date="1467720000">- update to 1.9.11 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.12-1" date="1467720001">- update to 1.9.12 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.13-1" date="1467720002">- update to 1.9.13 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.14-1" date="1467720003">- update to 1.9.14 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.15-1" date="1467720004">- update to 1.9.15 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.0-1" date="1467720005">- update to 1.10.0
- update nginx sticky module ng to version 1.2.6
- update nginx push stream module plugin to version 0.5.2
- update naxsi plugin to version 0.54
- enabled stream module</changelog>
</package>
<package pkgid="31107487d6c90d44dceee52333c3f48d78133d2925303e69f37593b11e2b4ea9" name="nginx" arch="src">
  <version epoch="0" ver="1.10.1" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.8-1" date="1449835200">- update to 1.9.8 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.9-1" date="1449835201">- update to 1.9.9 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.10-1" date="1455537600">- update to 1.9.10 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.11-1" date="1467720000">- update to 1.9.11 (mainline version)
- added dynamic modules support
- added module nginx-devel-kit (modular supported)
- added module nginx-form-input (modular supported)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.12-1" date="1467720001">- update to 1.9.12 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.13-1" date="1467720002">- update to 1.9.13 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.14-1" date="1467720003">- update to 1.9.14 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.15-1" date="1467720004">- update to 1.9.15 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.0-1" date="1467720005">- update to 1.10.0
- update nginx sticky module ng to version 1.2.6
- update nginx push stream module plugin to version 0.5.2
- update naxsi plugin to version 0.55rc2
- changed nginx push stream module to modular support
- changed nginx naxsi module to modular support
- enabled stream module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.1-1" date="1467979200">- update to 1.10.1
- added a patch to sticky module ng to support modular mode
- added a patch to upstream fair to support modular mode
- changed internal NGiNX modules to modular mode
- added echo module version 0.59
- added rtmp module version 1.1.8</changelog>
</package>
<package pkgid="d2b63de682e6eec0f6aab5f790f2515705fc8d26459b2ff4de9721b157508e15" name="nginx" arch="src">
  <version epoch="0" ver="1.10.2" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.9-1" date="1449835200">- update to 1.9.9 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.10-1" date="1455537600">- update to 1.9.10 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.11-1" date="1467720000">- update to 1.9.11 (mainline version)
- added dynamic modules support
- added module nginx-devel-kit (modular supported)
- added module nginx-form-input (modular supported)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.12-1" date="1467720001">- update to 1.9.12 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.13-1" date="1467720002">- update to 1.9.13 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.14-1" date="1467720003">- update to 1.9.14 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.15-1" date="1467720004">- update to 1.9.15 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.0-1" date="1467720005">- update to 1.10.0
- update nginx sticky module ng to version 1.2.6
- update nginx push stream module plugin to version 0.5.2
- update naxsi plugin to version 0.55rc2
- changed nginx push stream module to modular support
- changed nginx naxsi module to modular support
- enabled stream module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.1-1" date="1467979200">- update to 1.10.1
- added a patch to sticky module ng to support modular mode
- added a patch to upstream fair to support modular mode
- changed internal NGiNX modules to modular mode
- added echo module version 0.59
- added rtmp module version 1.1.8</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.2-1" date="1477915200">- update to 1.10.2</changelog>
</package>
<package pkgid="2a92ed1c8b72ea03c7f179e5addfc430eeb9de157941dfe414935d0abe45ac67" name="nginx" arch="src">
  <version epoch="0" ver="1.10.3" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.10-1" date="1455537600">- update to 1.9.10 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.11-1" date="1467720000">- update to 1.9.11 (mainline version)
- added dynamic modules support
- added module nginx-devel-kit (modular supported)
- added module nginx-form-input (modular supported)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.12-1" date="1467720001">- update to 1.9.12 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.13-1" date="1467720002">- update to 1.9.13 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.14-1" date="1467720003">- update to 1.9.14 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.15-1" date="1467720004">- update to 1.9.15 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.0-1" date="1467720005">- update to 1.10.0
- update nginx sticky module ng to version 1.2.6
- update nginx push stream module plugin to version 0.5.2
- update naxsi plugin to version 0.55rc2
- changed nginx push stream module to modular support
- changed nginx naxsi module to modular support
- enabled stream module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.1-1" date="1467979200">- update to 1.10.1
- added a patch to sticky module ng to support modular mode
- added a patch to upstream fair to support modular mode
- changed internal NGiNX modules to modular mode
- added echo module version 0.59
- added rtmp module version 1.1.8</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.2-1" date="1477915200">- update to 1.10.2</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.3-1" date="1485864000">- update to 1.10.3</changelog>
</package>
<package pkgid="6a5b1fafe655f0b4c23576a7dfce884eff67556ba7b9333f21890d64bf8d6270" name="nginx" arch="src">
  <version epoch="0" ver="1.11.0" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.7-1" date="1449835200">- update to 1.9.7 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.8-1" date="1449835201">- update to 1.9.8 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.9-1" date="1449835202">- update to 1.9.9 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.10-1" date="1455537600">- update to 1.9.10 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.11-1" date="1467720000">- update to 1.9.11 (mainline version)
- added dynamic modules support
- added module nginx-devel-kit (modular supported)
- added module nginx-form-input (modular supported)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.12-1" date="1467720001">- update to 1.9.12 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.13-1" date="1467720002">- update to 1.9.13 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.14-1" date="1467720003">- update to 1.9.14 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.15-1" date="1467720004">- update to 1.9.15 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.11.0-1" date="1467720005">- update to 1.11.0 (mainline version)
- update nginx sticky module ng to version 1.2.6
- update nginx push stream module plugin to version 0.5.2
- update naxsi plugin to version 0.54
- changed nginx push stream module to modular support
- enabled stream module</changelog>
</package>
<package pkgid="bd22c2f7507041d5a2e736a6755a519b2ed5eebab22c8abccebcb228b12149ba" name="nginx" arch="src">
  <version epoch="0" ver="1.11.1" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.8-1" date="1449835200">- update to 1.9.8 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.9-1" date="1449835201">- update to 1.9.9 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.10-1" date="1455537600">- update to 1.9.10 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.11-1" date="1467720000">- update to 1.9.11 (mainline version)
- added dynamic modules support
- added module nginx-devel-kit (modular supported)
- added module nginx-form-input (modular supported)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.12-1" date="1467720001">- update to 1.9.12 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.13-1" date="1467720002">- update to 1.9.13 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.14-1" date="1467720003">- update to 1.9.14 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.15-1" date="1467720004">- update to 1.9.15 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.11.0-1" date="1467720005">- update to 1.11.0 (mainline version)
- update nginx sticky module ng to version 1.2.6
- update nginx push stream module plugin to version 0.5.2
- update naxsi plugin to version 0.54
- changed nginx push stream module to modular support
- enabled stream module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.11.1-1" date="1467720006">- update to 1.11.1 (mainline version)</changelog>
</package>
<package pkgid="f78e370d8da48b3172def7a54fec4470df93a9aeda9a1cef182a95176ca5e521" name="nginx" arch="src">
  <version epoch="0" ver="1.11.2" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.9-1" date="1449835200">- update to 1.9.9 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.10-1" date="1455537600">- update to 1.9.10 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.11-1" date="1467720000">- update to 1.9.11 (mainline version)
- added dynamic modules support
- added module nginx-devel-kit (modular supported)
- added module nginx-form-input (modular supported)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.12-1" date="1467720001">- update to 1.9.12 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.13-1" date="1467720002">- update to 1.9.13 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.14-1" date="1467720003">- update to 1.9.14 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.15-1" date="1467720004">- update to 1.9.15 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.11.0-1" date="1467720005">- update to 1.11.0 (mainline version)
- update nginx sticky module ng to version 1.2.6
- update nginx push stream module plugin to version 0.5.2
- update naxsi plugin to version 0.54
- changed nginx push stream module to modular support
- enabled stream module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.11.1-1" date="1467720006">- update to 1.11.1 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.11.2-1" date="1467720007">- update to 1.11.2 (mainline version)
- removed nginx sticky module ng due to incompatibility
- update naxsi plugin to version 0.55rc2
- changed nginx naxsi module to modular support</changelog>
</package>
<package pkgid="8ee4e66ae5d782f51c36a39235a603f269e43b904d9bf25e6466505df06ba090" name="nginx" arch="src">
  <version epoch="0" ver="1.12.0" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.11-1" date="1467720000">- update to 1.9.11 (mainline version)
- added dynamic modules support
- added module nginx-devel-kit (modular supported)
- added module nginx-form-input (modular supported)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.12-1" date="1467720001">- update to 1.9.12 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.13-1" date="1467720002">- update to 1.9.13 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.14-1" date="1467720003">- update to 1.9.14 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.15-1" date="1467720004">- update to 1.9.15 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.0-1" date="1467720005">- update to 1.10.0
- update nginx sticky module ng to version 1.2.6
- update nginx push stream module plugin to version 0.5.2
- update naxsi plugin to version 0.55rc2
- changed nginx push stream module to modular support
- changed nginx naxsi module to modular support
- enabled stream module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.1-1" date="1467979200">- update to 1.10.1
- added a patch to sticky module ng to support modular mode
- added a patch to upstream fair to support modular mode
- changed internal NGiNX modules to modular mode
- added echo module version 0.59
- added rtmp module version 1.1.8</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.2-1" date="1477915200">- update to 1.10.2</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.3-1" date="1485864000">- update to 1.10.3</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.0-1" date="1491998400">- update to 1.12.0
- added auth ldap module version 0.1
- update nginx echo module to version 0.60
- update naxsi plugin to version 0.55.3
- update rtmp module to version 1.1.11
- fixed the sticky module ng modular support
- fixed the echo module ngx_chain_t usage</changelog>
</package>
<package pkgid="f7999f13d1ba3a3b0e826111df332430c1afaa858e410ba9ef09bb882f6c539b" name="nginx" arch="src">
  <version epoch="0" ver="1.12.1" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.12-1" date="1467720000">- update to 1.9.12 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.13-1" date="1467720001">- update to 1.9.13 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.14-1" date="1467720002">- update to 1.9.14 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.15-1" date="1467720003">- update to 1.9.15 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.0-1" date="1467720004">- update to 1.10.0
- update nginx sticky module ng to version 1.2.6
- update nginx push stream module plugin to version 0.5.2
- update naxsi plugin to version 0.55rc2
- changed nginx push stream module to modular support
- changed nginx naxsi module to modular support
- enabled stream module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.1-1" date="1467979200">- update to 1.10.1
- added a patch to sticky module ng to support modular mode
- added a patch to upstream fair to support modular mode
- changed internal NGiNX modules to modular mode
- added echo module version 0.59
- added rtmp module version 1.1.8</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.2-1" date="1477915200">- update to 1.10.2</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.3-1" date="1485864000">- update to 1.10.3</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.0-1" date="1491998400">- update to 1.12.0
- added auth ldap module version 0.1
- update nginx echo module to version 0.60
- update naxsi plugin to version 0.55.3
- update rtmp module to version 1.1.11
- fixed the sticky module ng modular support
- fixed the echo module ngx_chain_t usage</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.1-1" date="1504526400">- update to 1.12.1
- update nginx rtmp module to version 1.2.0
- update echo nginx module to version 0.61</changelog>
</package>
<package pkgid="13d9d249b768c34e4498d03cecc80421943db333d5496f1380c4c7051e304435" name="nginx" arch="src">
  <version epoch="0" ver="1.12.2" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.13-1" date="1467720000">- update to 1.9.13 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.14-1" date="1467720001">- update to 1.9.14 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.15-1" date="1467720002">- update to 1.9.15 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.0-1" date="1467720003">- update to 1.10.0
- update nginx sticky module ng to version 1.2.6
- update nginx push stream module plugin to version 0.5.2
- update naxsi plugin to version 0.55rc2
- changed nginx push stream module to modular support
- changed nginx naxsi module to modular support
- enabled stream module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.1-1" date="1467979200">- update to 1.10.1
- added a patch to sticky module ng to support modular mode
- added a patch to upstream fair to support modular mode
- changed internal NGiNX modules to modular mode
- added echo module version 0.59
- added rtmp module version 1.1.8</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.2-1" date="1477915200">- update to 1.10.2</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.3-1" date="1485864000">- update to 1.10.3</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.0-1" date="1491998400">- update to 1.12.0
- added auth ldap module version 0.1
- update nginx echo module to version 0.60
- update naxsi plugin to version 0.55.3
- update rtmp module to version 1.1.11
- fixed the sticky module ng modular support
- fixed the echo module ngx_chain_t usage</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.1-1" date="1504526400">- update to 1.12.1
- update nginx rtmp module to version 1.2.0
- update echo nginx module to version 0.61</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.2-1" date="1509105600">- update to 1.12.1
- update nginx push stream module to version 0.5.4</changelog>
</package>
<package pkgid="5bdb40635c363dccebf437a5dab5b6364e097c4dc312e3586a9a551ad6b1a249" name="nginx" arch="src">
  <version epoch="0" ver="1.14.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.14-1" date="1467720000">- update to 1.9.14 (mainline version)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.15-1" date="1467720001">- update to 1.9.15 (mainline version)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.0-1" date="1467720002">- update to 1.10.0
- update nginx sticky module ng to version 1.2.6
- update nginx push stream module plugin to version 0.5.2
- update naxsi plugin to version 0.55rc2
- changed nginx push stream module to modular support
- changed nginx naxsi module to modular support
- enabled stream module</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.1-1" date="1467979200">- update to 1.10.1
- added a patch to sticky module ng to support modular mode
- added a patch to upstream fair to support modular mode
- changed internal NGiNX modules to modular mode
- added echo module version 0.59
- added rtmp module version 1.1.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.2-1" date="1477915200">- update to 1.10.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.3-1" date="1485864000">- update to 1.10.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.0-1" date="1491998400">- update to 1.12.0
- added auth ldap module version 0.1
- update nginx echo module to version 0.60
- update naxsi plugin to version 0.55.3
- update rtmp module to version 1.1.11
- fixed the sticky module ng modular support
- fixed the echo module ngx_chain_t usage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.1-1" date="1504526400">- update to 1.12.1
- update nginx rtmp module to version 1.2.0
- update echo nginx module to version 0.61</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.2-1" date="1509105600">- update to 1.12.1
- update nginx push stream module to version 0.5.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-1" date="1523966400">- update to 1.14.0
- update nginx rtmp module to version 1.2.1</changelog>
</package>
<package pkgid="f23db54588800a4c22e09f2176398351fc9dd3d31f0c17ac12d25b41c55d5f6e" name="nginx" arch="src">
  <version epoch="0" ver="1.14.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.15-1" date="1467720000">- update to 1.9.15 (mainline version)</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.0-1" date="1467720001">- update to 1.10.0
- update nginx sticky module ng to version 1.2.6
- update nginx push stream module plugin to version 0.5.2
- update naxsi plugin to version 0.55rc2
- changed nginx push stream module to modular support
- changed nginx naxsi module to modular support
- enabled stream module</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.1-1" date="1467979200">- update to 1.10.1
- added a patch to sticky module ng to support modular mode
- added a patch to upstream fair to support modular mode
- changed internal NGiNX modules to modular mode
- added echo module version 0.59
- added rtmp module version 1.1.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.2-1" date="1477915200">- update to 1.10.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.3-1" date="1485864000">- update to 1.10.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.0-1" date="1491998400">- update to 1.12.0
- added auth ldap module version 0.1
- update nginx echo module to version 0.60
- update naxsi plugin to version 0.55.3
- update rtmp module to version 1.1.11
- fixed the sticky module ng modular support
- fixed the echo module ngx_chain_t usage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.1-1" date="1504526400">- update to 1.12.1
- update nginx rtmp module to version 1.2.0
- update echo nginx module to version 0.61</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.2-1" date="1509105600">- update to 1.12.1
- update nginx push stream module to version 0.5.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-1" date="1523966400">- update to 1.14.0
- update nginx rtmp module to version 1.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-2" date="1541505600">- update naxsi plugin to version 0.56</changelog>
</package>
<package pkgid="f6b55e2a57cf5b8d5f0bbcd862f7fc87f209cea5358bbb4df3ea6902cb88b897" name="nginx" arch="src">
  <version epoch="0" ver="1.14.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.0-1" date="1467720000">- update to 1.10.0
- update nginx sticky module ng to version 1.2.6
- update nginx push stream module plugin to version 0.5.2
- update naxsi plugin to version 0.55rc2
- changed nginx push stream module to modular support
- changed nginx naxsi module to modular support
- enabled stream module</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.1-1" date="1467979200">- update to 1.10.1
- added a patch to sticky module ng to support modular mode
- added a patch to upstream fair to support modular mode
- changed internal NGiNX modules to modular mode
- added echo module version 0.59
- added rtmp module version 1.1.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.2-1" date="1477915200">- update to 1.10.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.3-1" date="1485864000">- update to 1.10.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.0-1" date="1491998400">- update to 1.12.0
- added auth ldap module version 0.1
- update nginx echo module to version 0.60
- update naxsi plugin to version 0.55.3
- update rtmp module to version 1.1.11
- fixed the sticky module ng modular support
- fixed the echo module ngx_chain_t usage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.1-1" date="1504526400">- update to 1.12.1
- update nginx rtmp module to version 1.2.0
- update echo nginx module to version 0.61</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.2-1" date="1509105600">- update to 1.12.1
- update nginx push stream module to version 0.5.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-1" date="1523966400">- update to 1.14.0
- update nginx rtmp module to version 1.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-2" date="1541505600">- update naxsi plugin to version 0.56</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.1-1" date="1541505601">- update to 1.14.1</changelog>
</package>
<package pkgid="55bef1b66350ca96f57030e8198df3cfd1d86f4ac5fe9b615109efdc3ad29a79" name="nginx" arch="src">
  <version epoch="0" ver="1.14.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.1-1" date="1467979200">- update to 1.10.1
- added a patch to sticky module ng to support modular mode
- added a patch to upstream fair to support modular mode
- changed internal NGiNX modules to modular mode
- added echo module version 0.59
- added rtmp module version 1.1.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.2-1" date="1477915200">- update to 1.10.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.3-1" date="1485864000">- update to 1.10.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.0-1" date="1491998400">- update to 1.12.0
- added auth ldap module version 0.1
- update nginx echo module to version 0.60
- update naxsi plugin to version 0.55.3
- update rtmp module to version 1.1.11
- fixed the sticky module ng modular support
- fixed the echo module ngx_chain_t usage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.1-1" date="1504526400">- update to 1.12.1
- update nginx rtmp module to version 1.2.0
- update echo nginx module to version 0.61</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.2-1" date="1509105600">- update to 1.12.1
- update nginx push stream module to version 0.5.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-1" date="1523966400">- update to 1.14.0
- update nginx rtmp module to version 1.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-2" date="1541505600">- update naxsi plugin to version 0.56</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.1-1" date="1541505601">- update to 1.14.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.2-1" date="1553601600">- update to 1.14.2</changelog>
</package>
<package pkgid="48207ee1f09de91e8d302d493d3c3f7a0beb3bcf85de960d5e0144bb16471ad0" name="nginx" arch="src">
  <version epoch="0" ver="1.16.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.2-1" date="1477915200">- update to 1.10.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.3-1" date="1485864000">- update to 1.10.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.0-1" date="1491998400">- update to 1.12.0
- added auth ldap module version 0.1
- update nginx echo module to version 0.60
- update naxsi plugin to version 0.55.3
- update rtmp module to version 1.1.11
- fixed the sticky module ng modular support
- fixed the echo module ngx_chain_t usage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.1-1" date="1504526400">- update to 1.12.1
- update nginx rtmp module to version 1.2.0
- update echo nginx module to version 0.61</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.2-1" date="1509105600">- update to 1.12.1
- update nginx push stream module to version 0.5.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-1" date="1523966400">- update to 1.14.0
- update nginx rtmp module to version 1.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-2" date="1541505600">- update naxsi plugin to version 0.56</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.1-1" date="1541505601">- update to 1.14.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.2-1" date="1553601600">- update to 1.14.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.0-1" date="1563278400">- update to 1.16.0</changelog>
</package>
<package pkgid="e7bfb79e745334b82528814fc94bfcd1f4df6fca73569243e7a47622849434a7" name="nginx" arch="src">
  <version epoch="0" ver="1.16.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.10.3-1" date="1485864000">- update to 1.10.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.0-1" date="1491998400">- update to 1.12.0
- added auth ldap module version 0.1
- update nginx echo module to version 0.60
- update naxsi plugin to version 0.55.3
- update rtmp module to version 1.1.11
- fixed the sticky module ng modular support
- fixed the echo module ngx_chain_t usage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.1-1" date="1504526400">- update to 1.12.1
- update nginx rtmp module to version 1.2.0
- update echo nginx module to version 0.61</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.2-1" date="1509105600">- update to 1.12.1
- update nginx push stream module to version 0.5.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-1" date="1523966400">- update to 1.14.0
- update nginx rtmp module to version 1.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-2" date="1541505600">- update naxsi plugin to version 0.56</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.1-1" date="1541505601">- update to 1.14.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.2-1" date="1553601600">- update to 1.14.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.0-1" date="1563278400">- update to 1.16.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.1-1" date="1568721600">- update to 1.16.1</changelog>
</package>
<package pkgid="9f4f2ce19b6d9f1e5ff776a4d4eb3cebcf2939038e4612e7c1eaffbd6b254d43" name="nginx" arch="src">
  <version epoch="0" ver="1.18.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.0-1" date="1491998400">- update to 1.12.0
- added auth ldap module version 0.1
- update nginx echo module to version 0.60
- update naxsi plugin to version 0.55.3
- update rtmp module to version 1.1.11
- fixed the sticky module ng modular support
- fixed the echo module ngx_chain_t usage</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.1-1" date="1504526400">- update to 1.12.1
- update nginx rtmp module to version 1.2.0
- update echo nginx module to version 0.61</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.2-1" date="1509105600">- update to 1.12.1
- update nginx push stream module to version 0.5.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-1" date="1523966400">- update to 1.14.0
- update nginx rtmp module to version 1.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-2" date="1541505600">- update naxsi plugin to version 0.56</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.1-1" date="1541505601">- update to 1.14.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.2-1" date="1553601600">- update to 1.14.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.0-1" date="1563278400">- update to 1.16.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.1-1" date="1568721600">- update to 1.16.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.18.0-1" date="1596801600">- update to 1.18.0
- update echo nginx module to version 0.62
- update devel kit module to version 0.3.1</changelog>
</package>
<package pkgid="81259eb7072a9227468494e0e817113d26cdc6e26fccac48ff5efb1242ab95eb" name="nginx" arch="src">
  <version epoch="0" ver="1.20.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.1-1" date="1504526400">- update to 1.12.1
- update nginx rtmp module to version 1.2.0
- update echo nginx module to version 0.61</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.2-1" date="1509105600">- update to 1.12.1
- update nginx push stream module to version 0.5.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-1" date="1523966400">- update to 1.14.0
- update nginx rtmp module to version 1.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-2" date="1541505600">- update naxsi plugin to version 0.56</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.1-1" date="1541505601">- update to 1.14.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.2-1" date="1553601600">- update to 1.14.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.0-1" date="1563278400">- update to 1.16.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.1-1" date="1568721600">- update to 1.16.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.18.0-1" date="1596801600">- update to 1.18.0
- update echo nginx module to version 0.62
- update devel kit module to version 0.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.0-1" date="1620388800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
</package>
<package pkgid="3d0aead63b12073607ecf816845e2b7515f0c6fd74a52d08d49f244d77b584d8" name="nginx" arch="src">
  <version epoch="0" ver="1.20.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.12.2-1" date="1509105600">- update to 1.12.1
- update nginx push stream module to version 0.5.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-1" date="1523966400">- update to 1.14.0
- update nginx rtmp module to version 1.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-2" date="1541505600">- update naxsi plugin to version 0.56</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.1-1" date="1541505601">- update to 1.14.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.2-1" date="1553601600">- update to 1.14.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.0-1" date="1563278400">- update to 1.16.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.1-1" date="1568721600">- update to 1.16.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.18.0-1" date="1596801600">- update to 1.18.0
- update echo nginx module to version 0.62
- update devel kit module to version 0.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.0-1" date="1620388800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-1" date="1632139200">- update to 1.20.1</changelog>
</package>
<package pkgid="9e08deaaa143f6ec5af0b3fd083f8bb5a05e4fefcbfdd49035377a98894c254f" name="nginx" arch="src">
  <version epoch="0" ver="1.20.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-1" date="1523966400">- update to 1.14.0
- update nginx rtmp module to version 1.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-2" date="1541505600">- update naxsi plugin to version 0.56</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.1-1" date="1541505601">- update to 1.14.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.2-1" date="1553601600">- update to 1.14.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.0-1" date="1563278400">- update to 1.16.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.1-1" date="1568721600">- update to 1.16.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.18.0-1" date="1596801600">- update to 1.18.0
- update echo nginx module to version 0.62
- update devel kit module to version 0.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.0-1" date="1620388800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-1" date="1632139200">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-2" date="1642161600">- added njs plugin at version 0.7.1</changelog>
</package>
<package pkgid="fe9455459b127dcfba696db9bb5249cf847a1cc6fd19d93933f75fa7864df821" name="nginx" arch="src">
  <version epoch="0" ver="1.20.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.0-2" date="1541505600">- update naxsi plugin to version 0.56</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.1-1" date="1541505601">- update to 1.14.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.2-1" date="1553601600">- update to 1.14.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.0-1" date="1563278400">- update to 1.16.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.1-1" date="1568721600">- update to 1.16.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.18.0-1" date="1596801600">- update to 1.18.0
- update echo nginx module to version 0.62
- update devel kit module to version 0.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.0-1" date="1620388800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-1" date="1632139200">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-2" date="1642161600">- added njs plugin at version 0.7.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-3" date="1644235200">- added headers more plugin at version 0.33</changelog>
</package>
<package pkgid="36eab85353f3d458a42dfed24de3f59112ef04247c60106dfc0ffcf6c4995c64" name="nginx" arch="src">
  <version epoch="0" ver="1.20.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.1-1" date="1541505600">- update to 1.14.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.2-1" date="1553601600">- update to 1.14.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.0-1" date="1563278400">- update to 1.16.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.1-1" date="1568721600">- update to 1.16.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.18.0-1" date="1596801600">- update to 1.18.0
- update echo nginx module to version 0.62
- update devel kit module to version 0.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.0-1" date="1620388800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-1" date="1632139200">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-2" date="1642161600">- added njs plugin at version 0.7.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-3" date="1644235200">- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
</package>
<package pkgid="dee1eacd798923bb9a83671954e5e669e97f7aa66d5412b31ca72d2f90a112fc" name="nginx" arch="src">
  <version epoch="0" ver="1.20.2" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.14.2-1" date="1553601600">- update to 1.14.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.0-1" date="1563278400">- update to 1.16.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.1-1" date="1568721600">- update to 1.16.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.18.0-1" date="1596801600">- update to 1.18.0
- update echo nginx module to version 0.62
- update devel kit module to version 0.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.0-1" date="1620388800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-1" date="1632139200">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-2" date="1642161600">- added njs plugin at version 0.7.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-3" date="1644235200">- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
</package>
<package pkgid="34589ed088c29122b786f5001d8f37fc621b1b1e11b95ee4733d628af55fbf7b" name="nginx" arch="src">
  <version epoch="0" ver="1.22.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.0-1" date="1563278400">- update to 1.16.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.1-1" date="1568721600">- update to 1.16.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.18.0-1" date="1596801600">- update to 1.18.0
- update echo nginx module to version 0.62
- update devel kit module to version 0.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.0-1" date="1620388800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-1" date="1632139200">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-2" date="1642161600">- added njs plugin at version 0.7.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-3" date="1644235200">- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
</package>
<package pkgid="1039dea7b37f0f09f650a45e7bdd2b374b78d1dbdc04c0a5de84cbcabb7017d1" name="nginx" arch="src">
  <version epoch="0" ver="1.22.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.16.1-1" date="1568721600">- update to 1.16.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.18.0-1" date="1596801600">- update to 1.18.0
- update echo nginx module to version 0.62
- update devel kit module to version 0.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.0-1" date="1620388800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-1" date="1632139200">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-2" date="1642161600">- added njs plugin at version 0.7.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-3" date="1644235200">- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
</package>
<package pkgid="6a2788a6ca525ec5368a63d15743a495185ecc4eed192b98ddde5fd84dd79e03" name="nginx" arch="src">
  <version epoch="0" ver="1.22.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.18.0-1" date="1596801600">- update to 1.18.0
- update echo nginx module to version 0.62
- update devel kit module to version 0.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.0-1" date="1620388800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-1" date="1632139200">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-2" date="1642161600">- added njs plugin at version 0.7.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-3" date="1644235200">- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
</package>
<package pkgid="63f2047c4095a5a288cb445cc06a7859194ca705b904fa45ba66d19d53a35bff" name="nginx" arch="src">
  <version epoch="0" ver="1.22.1" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.0-1" date="1620388800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-1" date="1632139200">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-2" date="1642161600">- added njs plugin at version 0.7.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-3" date="1644235200">- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-2" date="1669291200">- update ngx_devel_kit to version 0.3.2</changelog>
</package>
<package pkgid="e5fa34a6ee448334197266bd08a12bb7e1e891c9db7786289c03f51758dd3b8d" name="nginx" arch="src">
  <version epoch="0" ver="1.22.1" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-1" date="1632139200">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-2" date="1642161600">- added njs plugin at version 0.7.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-3" date="1644235200">- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-2" date="1669291200">- update ngx_devel_kit to version 0.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-3" date="1708689600">- added lua module version 0.10.26</changelog>
</package>
<package pkgid="80930a70b14b51598b9d2d26627221acffdc93ebb5c070528dbced561b6cda23" name="nginx" arch="src">
  <version epoch="0" ver="1.22.1" rel="4.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-2" date="1642161600">- added njs plugin at version 0.7.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.1-3" date="1644235200">- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.20.2-1" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-2" date="1669291200">- update ngx_devel_kit to version 0.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-3" date="1708689600">- added lua module version 0.10.26</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-4" date="1709553600">- added lua ipc module version 0.1.6</changelog>
</package>
<package pkgid="aeb242c45cd6427c650e1c1aeccda526e56a5768bae39cb311b6ab66133ddbc5" name="nginx" arch="src">
  <version epoch="1" ver="1.24.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.18.0-4" date="1636200000">- Added http push stream module version 0.5.4
- Added http sticky ng module version 1.2.6
- Added http naxsi module version 0.56
- Added devel kit module version 0.3.1
- Added http form input module version 0.12
- Added http echo module version 0.62
- Added rtmp module version 1.2.1
- Added http authenticate ldap module version 0.1
- Added http upstream fair module version 0.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-0-1" date="1636372800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-1-1" date="1636372801">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.1-2" date="1644235200">- added njs plugin at version 0.7.1
- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
</package>
<package pkgid="72531e3ade6f9ff21f43917077c0c79eb7fc18b890b131351f590c4dddc479fa" name="nginx" arch="src">
  <version epoch="1" ver="1.26.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-0-1" date="1636372800">- update to 1.20.0
- update naxsi module to version 1.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-1-1" date="1636372801">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.1-2" date="1644235200">- added njs plugin at version 0.7.1
- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
</package>
<package pkgid="3808745a3c08ca506893968fc46c03f354954c9e5a4cca2a10ba914e97b8145b" name="nginx" arch="src">
  <version epoch="1" ver="1.26.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20-1-1" date="1636372800">- update to 1.20.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.1-2" date="1644235200">- added njs plugin at version 0.7.1
- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
</package>
<package pkgid="13bdf4965e2086a8c4fb15e1d2db797876112fbd2b758d54e6c44bb1bef5e2f5" name="nginx" arch="src">
  <version epoch="1" ver="1.26.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.1-2" date="1644235200">- added njs plugin at version 0.7.1
- added headers more plugin at version 0.33</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.2-1" date="1723464000">- update to 1.26.2</changelog>
</package>
<package pkgid="e5b1adefe972704f8729fc102688c5ac82e16a0fb374ad364df90ba84681a446" name="nginx" arch="src">
  <version epoch="1" ver="1.26.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-1" date="1650283200">- update to 1.20.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283201">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.2-1" date="1723464000">- update to 1.26.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.3-1" date="1738756800">- update to 1.26.3</changelog>
</package>
<package pkgid="978b75ca73d8a06df8fec7b8045060fbc9474b5dd50ae23c560ccb7dea0aac1c" name="nginx" arch="src">
  <version epoch="1" ver="1.28.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1:1.20.2-2" date="1650283200">- update nginx push stream module to version 0.5.5
- update nginx rtmp module to version 1.2.2
- update njs module to version 0.7.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.2-1" date="1723464000">- update to 1.26.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.3-1" date="1738756800">- update to 1.26.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.0-1" date="1745409600">- update to 1.28.0</changelog>
</package>
<package pkgid="c64a77db8185773497e38e278908e2420c6a89fd894210d6e68af61dcc55071b" name="nginx" arch="src">
  <version epoch="1" ver="1.28.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-1" date="1660219200">- update to 1.22.0
- update njs module to version 0.7.6
- update headers more module to version 0.34</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.2-1" date="1723464000">- update to 1.26.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.3-1" date="1738756800">- update to 1.26.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.0-1" date="1745409600">- update to 1.28.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.1-1" date="1766491200">- update to 1.28.1</changelog>
</package>
<package pkgid="86cccabb1819efdd4770ff48d04fa755d32926b54a54100ddc00c9693d30fead" name="nginx" arch="src">
  <version epoch="1" ver="1.28.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.0-2" date="1661428800">- added http proxy connect module version 0.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.2-1" date="1723464000">- update to 1.26.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.3-1" date="1738756800">- update to 1.26.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.0-1" date="1745409600">- update to 1.28.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.1-1" date="1766491200">- update to 1.28.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.2-1" date="1770206400">- update to 1.28.2
- update njs module to version 0.9.1</changelog>
</package>
<package pkgid="3c53de1df5a55cc4cef3f8537345834fc9c639c4871d5803ab0fa32ab4031a91" name="nginx" arch="src">
  <version epoch="1" ver="1.28.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.22.1-1" date="1666180800">- update to 1.22.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.2-1" date="1723464000">- update to 1.26.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.3-1" date="1738756800">- update to 1.26.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.0-1" date="1745409600">- update to 1.28.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.1-1" date="1766491200">- update to 1.28.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.2-1" date="1770206400">- update to 1.28.2
- update njs module to version 0.9.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.3-1" date="1774353600">- update to 1.28.3</changelog>
</package>
<package pkgid="fcbd8e0864e15dc705dd2a5db802d0919610c445e723083b76330ff3eb76a597" name="nginx" arch="src">
  <version epoch="1" ver="1.30.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.24.0-1" date="1681128000">- update njs module to version 0.9.1
- update naxsi module to version 1.7
- update nginx-push-stream-module module to version 0.6.0
- update to 1.24.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.2-1" date="1723464000">- update to 1.26.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.3-1" date="1738756800">- update to 1.26.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.0-1" date="1745409600">- update to 1.28.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.1-1" date="1766491200">- update to 1.28.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.2-1" date="1770206400">- update to 1.28.2
- update njs module to version 0.9.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.3-1" date="1774353600">- update to 1.28.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.30.0-1" date="1776168000">- update to 1.30.0
- update http-proxy-connect-module to version 0.0.7</changelog>
</package>
<package pkgid="a6b5407804e1ff5a4ced51c910803efe68cf23e98a6e29b67fe9df63e5ad991c" name="nginx" arch="src">
  <version epoch="1" ver="1.30.0" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.0-1" date="1713873600">- update to 1.26.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.2-1" date="1723464000">- update to 1.26.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.3-1" date="1738756800">- update to 1.26.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.0-1" date="1745409600">- update to 1.28.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.1-1" date="1766491200">- update to 1.28.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.2-1" date="1770206400">- update to 1.28.2
- update njs module to version 0.9.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.3-1" date="1774353600">- update to 1.28.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.30.0-1" date="1776168000">- update to 1.30.0
- update http-proxy-connect-module to version 0.0.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.30.0-2" date="1776945600">- update njs module to version 0.9.6</changelog>
</package>
<package pkgid="d391089f70e24843b75a8dda31831a910914190b49552e433d221c5036c731e1" name="nginx" arch="src">
  <version epoch="1" ver="1.30.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.1-1" date="1716984000">- update to 1.26.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.2-1" date="1723464000">- update to 1.26.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.3-1" date="1738756800">- update to 1.26.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.0-1" date="1745409600">- update to 1.28.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.1-1" date="1766491200">- update to 1.28.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.2-1" date="1770206400">- update to 1.28.2
- update njs module to version 0.9.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.3-1" date="1774353600">- update to 1.28.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.30.0-1" date="1776168000">- update to 1.30.0
- update http-proxy-connect-module to version 0.0.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.30.0-2" date="1776945600">- update njs module to version 0.9.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.30.1-1" date="1778673600">- update to 1.30.1
- update njs module to version 0.9.9</changelog>
</package>
<package pkgid="143d133a9f3421f7bda26e6cfbd8186e3aec477f494b9c100754150aa85d3042" name="nginx" arch="src">
  <version epoch="1" ver="1.30.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.2-1" date="1723464000">- update to 1.26.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.26.3-1" date="1738756800">- update to 1.26.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.0-1" date="1745409600">- update to 1.28.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.1-1" date="1766491200">- update to 1.28.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.2-1" date="1770206400">- update to 1.28.2
- update njs module to version 0.9.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.28.3-1" date="1774353600">- update to 1.28.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.30.0-1" date="1776168000">- update to 1.30.0
- update http-proxy-connect-module to version 0.0.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.30.0-2" date="1776945600">- update njs module to version 0.9.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.30.1-1" date="1778673600">- update to 1.30.1
- update njs module to version 0.9.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.30.2-1" date="1779451200">- update to 1.30.2</changelog>
</package>
<package pkgid="7b9961f407695366a6d786a203acb45647eb5a499198c080a980a29a5080c706" name="nginx" arch="src">
  <version epoch="0" ver="1.8.0" rel="3.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.5-1" date="1394712000">- update to 1.4.5</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1395144000">- update to 1.4.6</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1395144001">- update to 1.4.7</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.0-1" date="1399377600">- update to 1.6.0
- added linking parameters to fix CentOS/RHEL 7 linking bug</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.1-1" date="1413028800">- update to 1.6.1</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.2-1" date="1413028801">- update to 1.6.2</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.3-1" date="1429358400">- update to 1.6.3</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-1" date="1429704000">- update to 1.8.0</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-2" date="1430913600">- updated nginx-push-stream-module to version 0.5.1
- removed obsolete nginx-sticky-module
- added nginx-sticky-module-ng version 1.2.5</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-3" date="1432209600">- added support to systemd (RHEL/CentOS &gt;= 7)</changelog>
</package>
<package pkgid="9cdd929643cef65313ab846cc13fa0461f793a037a40453758d2aaaa66a8cd4d" name="nginx" arch="src">
  <version epoch="0" ver="1.8.1" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1395144000">- update to 1.4.6</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1395144001">- update to 1.4.7</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.0-1" date="1399377600">- update to 1.6.0
- added linking parameters to fix CentOS/RHEL 7 linking bug</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.1-1" date="1413028800">- update to 1.6.1</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.2-1" date="1413028801">- update to 1.6.2</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.3-1" date="1429358400">- update to 1.6.3</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-1" date="1429704000">- update to 1.8.0</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-2" date="1430913600">- updated nginx-push-stream-module to version 0.5.1
- removed obsolete nginx-sticky-module
- added nginx-sticky-module-ng version 1.2.5</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-3" date="1432209600">- added support to systemd (RHEL/CentOS &gt;= 7)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.1-1" date="1455537600">- update to 1.8.1</changelog>
</package>
<package pkgid="ef9034ad1c04de6e78a3b04500f482c14577316bc8c9844a1be77481669cde8d" name="nginx" arch="src">
  <version epoch="0" ver="1.9.0" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1395144000">- update to 1.4.6</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1395144001">- update to 1.4.7</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.0-1" date="1399377600">- update to 1.6.0
- added linking parameters to fix CentOS/RHEL 7 linking bug</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.1-1" date="1413028800">- update to 1.6.1</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.2-1" date="1413028801">- update to 1.6.2</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.3-1" date="1429358400">- update to 1.6.3</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-1" date="1429704000">- update to 1.8.0</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-2" date="1430913600">- updated nginx-push-stream-module to version 0.5.1
- removed obsolete nginx-sticky-module
- added nginx-sticky-module-ng version 1.2.5</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-3" date="1432209600">- added support to systemd (RHEL/CentOS &gt;= 7)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.0-1" date="1442923200">- update to 1.9.0 (mainline version)</changelog>
</package>
<package pkgid="eb48a15f4cb073f5a67740179edc438e83432ec1cc4f71bb88ac0379cb13b169" name="nginx" arch="src">
  <version epoch="0" ver="1.9.1" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1395144000">- update to 1.4.7</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.0-1" date="1399377600">- update to 1.6.0
- added linking parameters to fix CentOS/RHEL 7 linking bug</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.1-1" date="1413028800">- update to 1.6.1</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.2-1" date="1413028801">- update to 1.6.2</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.3-1" date="1429358400">- update to 1.6.3</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-1" date="1429704000">- update to 1.8.0</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-2" date="1430913600">- updated nginx-push-stream-module to version 0.5.1
- removed obsolete nginx-sticky-module
- added nginx-sticky-module-ng version 1.2.5</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-3" date="1432209600">- added support to systemd (RHEL/CentOS &gt;= 7)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.0-1" date="1442923200">- update to 1.9.0 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.1-1" date="1442923201">- update to 1.9.1 (mainline version)</changelog>
</package>
<package pkgid="26091186a51c56aff8f0deabb1d65fcb2bdd41cbc1c1a842604037ff0861ec55" name="nginx" arch="src">
  <version epoch="0" ver="1.9.10" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.1-1" date="1442923200">- update to 1.9.1 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.2-1" date="1442923201">- update to 1.9.2 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.3-1" date="1442923202">- update to 1.9.3 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.4-1" date="1442923203">- update to 1.9.4 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.5-1" date="1442923204">- update to 1.9.5 (mainline version)
- removed obsolete spdy module
- added experimental http2 module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.6-1" date="1449835200">- update to 1.9.6 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.7-1" date="1449835201">- update to 1.9.7 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.8-1" date="1449835202">- update to 1.9.8 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.9-1" date="1449835203">- update to 1.9.9 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.10-1" date="1455537600">- update to 1.9.10 (mainline version)</changelog>
</package>
<package pkgid="8f006cea41278c50d88d2a3ba5c0af907b4fde80499572eb4e1d68e026ab1174" name="nginx" arch="src">
  <version epoch="0" ver="1.9.11" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.2-1" date="1442923200">- update to 1.9.2 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.3-1" date="1442923201">- update to 1.9.3 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.4-1" date="1442923202">- update to 1.9.4 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.5-1" date="1442923203">- update to 1.9.5 (mainline version)
- removed obsolete spdy module
- added experimental http2 module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.6-1" date="1449835200">- update to 1.9.6 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.7-1" date="1449835201">- update to 1.9.7 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.8-1" date="1449835202">- update to 1.9.8 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.9-1" date="1449835203">- update to 1.9.9 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.10-1" date="1455537600">- update to 1.9.10 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.11-1" date="1467720000">- update to 1.9.11 (mainline version)
- added dynamic modules support
- added module nginx-devel-kit (modular supported)
- added module nginx-form-input (modular supported)</changelog>
</package>
<package pkgid="8c0a8267a4466fc775b34f0c8a0febc3c98469cc8d99ff1281ac14826e03caa2" name="nginx" arch="src">
  <version epoch="0" ver="1.9.12" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.3-1" date="1442923200">- update to 1.9.3 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.4-1" date="1442923201">- update to 1.9.4 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.5-1" date="1442923202">- update to 1.9.5 (mainline version)
- removed obsolete spdy module
- added experimental http2 module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.6-1" date="1449835200">- update to 1.9.6 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.7-1" date="1449835201">- update to 1.9.7 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.8-1" date="1449835202">- update to 1.9.8 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.9-1" date="1449835203">- update to 1.9.9 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.10-1" date="1455537600">- update to 1.9.10 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.11-1" date="1467720000">- update to 1.9.11 (mainline version)
- added dynamic modules support
- added module nginx-devel-kit (modular supported)
- added module nginx-form-input (modular supported)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.12-1" date="1467720001">- update to 1.9.12 (mainline version)</changelog>
</package>
<package pkgid="c1d176ec40e251bd52e4a3b731c19f4b2ec92d8ba112eaa023b06f3d5954c14f" name="nginx" arch="src">
  <version epoch="0" ver="1.9.13" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.4-1" date="1442923200">- update to 1.9.4 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.5-1" date="1442923201">- update to 1.9.5 (mainline version)
- removed obsolete spdy module
- added experimental http2 module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.6-1" date="1449835200">- update to 1.9.6 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.7-1" date="1449835201">- update to 1.9.7 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.8-1" date="1449835202">- update to 1.9.8 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.9-1" date="1449835203">- update to 1.9.9 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.10-1" date="1455537600">- update to 1.9.10 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.11-1" date="1467720000">- update to 1.9.11 (mainline version)
- added dynamic modules support
- added module nginx-devel-kit (modular supported)
- added module nginx-form-input (modular supported)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.12-1" date="1467720001">- update to 1.9.12 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.13-1" date="1467720002">- update to 1.9.13 (mainline version)</changelog>
</package>
<package pkgid="84a4739ae96183656d14e636047dfd92b7e8658909734a5d2266780ee83cc967" name="nginx" arch="src">
  <version epoch="0" ver="1.9.14" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.5-1" date="1442923200">- update to 1.9.5 (mainline version)
- removed obsolete spdy module
- added experimental http2 module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.6-1" date="1449835200">- update to 1.9.6 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.7-1" date="1449835201">- update to 1.9.7 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.8-1" date="1449835202">- update to 1.9.8 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.9-1" date="1449835203">- update to 1.9.9 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.10-1" date="1455537600">- update to 1.9.10 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.11-1" date="1467720000">- update to 1.9.11 (mainline version)
- added dynamic modules support
- added module nginx-devel-kit (modular supported)
- added module nginx-form-input (modular supported)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.12-1" date="1467720001">- update to 1.9.12 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.13-1" date="1467720002">- update to 1.9.13 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.14-1" date="1467720003">- update to 1.9.14 (mainline version)</changelog>
</package>
<package pkgid="40ba08f2b4bcfe44fbba17d2bdd59aa40b63740a6e836667bbb4aaa87bb697f8" name="nginx" arch="src">
  <version epoch="0" ver="1.9.15" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.6-1" date="1449835200">- update to 1.9.6 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.7-1" date="1449835201">- update to 1.9.7 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.8-1" date="1449835202">- update to 1.9.8 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.9-1" date="1449835203">- update to 1.9.9 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.10-1" date="1455537600">- update to 1.9.10 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.11-1" date="1467720000">- update to 1.9.11 (mainline version)
- added dynamic modules support
- added module nginx-devel-kit (modular supported)
- added module nginx-form-input (modular supported)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.12-1" date="1467720001">- update to 1.9.12 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.13-1" date="1467720002">- update to 1.9.13 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.14-1" date="1467720003">- update to 1.9.14 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.15-1" date="1467720004">- update to 1.9.15 (mainline version)</changelog>
</package>
<package pkgid="96770e7671f7b0cdcd3ee34267afd4b1cfd6a1cdae9dc1f68682873f58639ab5" name="nginx" arch="src">
  <version epoch="0" ver="1.9.2" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.0-1" date="1399377600">- update to 1.6.0
- added linking parameters to fix CentOS/RHEL 7 linking bug</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.1-1" date="1413028800">- update to 1.6.1</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.2-1" date="1413028801">- update to 1.6.2</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.3-1" date="1429358400">- update to 1.6.3</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-1" date="1429704000">- update to 1.8.0</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-2" date="1430913600">- updated nginx-push-stream-module to version 0.5.1
- removed obsolete nginx-sticky-module
- added nginx-sticky-module-ng version 1.2.5</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-3" date="1432209600">- added support to systemd (RHEL/CentOS &gt;= 7)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.0-1" date="1442923200">- update to 1.9.0 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.1-1" date="1442923201">- update to 1.9.1 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.2-1" date="1442923202">- update to 1.9.2 (mainline version)</changelog>
</package>
<package pkgid="e97b5ead3d1b83de96a7dfa9988a29141d6a1ee7782728adaec57ef5ec8f1717" name="nginx" arch="src">
  <version epoch="0" ver="1.9.3" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.1-1" date="1413028800">- update to 1.6.1</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.2-1" date="1413028801">- update to 1.6.2</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.3-1" date="1429358400">- update to 1.6.3</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-1" date="1429704000">- update to 1.8.0</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-2" date="1430913600">- updated nginx-push-stream-module to version 0.5.1
- removed obsolete nginx-sticky-module
- added nginx-sticky-module-ng version 1.2.5</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-3" date="1432209600">- added support to systemd (RHEL/CentOS &gt;= 7)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.0-1" date="1442923200">- update to 1.9.0 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.1-1" date="1442923201">- update to 1.9.1 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.2-1" date="1442923202">- update to 1.9.2 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.3-1" date="1442923203">- update to 1.9.3 (mainline version)</changelog>
</package>
<package pkgid="effb40cc42839cc246d092221a17d583ec4407c7f6bec1365eec6310d9e1450e" name="nginx" arch="src">
  <version epoch="0" ver="1.9.4" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.2-1" date="1413028800">- update to 1.6.2</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.3-1" date="1429358400">- update to 1.6.3</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-1" date="1429704000">- update to 1.8.0</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-2" date="1430913600">- updated nginx-push-stream-module to version 0.5.1
- removed obsolete nginx-sticky-module
- added nginx-sticky-module-ng version 1.2.5</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-3" date="1432209600">- added support to systemd (RHEL/CentOS &gt;= 7)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.0-1" date="1442923200">- update to 1.9.0 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.1-1" date="1442923201">- update to 1.9.1 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.2-1" date="1442923202">- update to 1.9.2 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.3-1" date="1442923203">- update to 1.9.3 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.4-1" date="1442923204">- update to 1.9.4 (mainline version)</changelog>
</package>
<package pkgid="6934e388fc4f704fdf2a9b1ba712338f935273dbc848f4efbd2056fe1a164637" name="nginx" arch="src">
  <version epoch="0" ver="1.9.5" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.3-1" date="1429358400">- update to 1.6.3</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-1" date="1429704000">- update to 1.8.0</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-2" date="1430913600">- updated nginx-push-stream-module to version 0.5.1
- removed obsolete nginx-sticky-module
- added nginx-sticky-module-ng version 1.2.5</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-3" date="1432209600">- added support to systemd (RHEL/CentOS &gt;= 7)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.0-1" date="1442923200">- update to 1.9.0 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.1-1" date="1442923201">- update to 1.9.1 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.2-1" date="1442923202">- update to 1.9.2 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.3-1" date="1442923203">- update to 1.9.3 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.4-1" date="1442923204">- update to 1.9.4 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.5-1" date="1442923205">- update to 1.9.5 (mainline version)
- removed obsolete spdy module
- added experimental http2 module</changelog>
</package>
<package pkgid="6c8b53d174b2bf440796614d71e48cb13d4bb8af56a26eba6b51f2e665f76165" name="nginx" arch="src">
  <version epoch="0" ver="1.9.6" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-1" date="1429704000">- update to 1.8.0</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-2" date="1430913600">- updated nginx-push-stream-module to version 0.5.1
- removed obsolete nginx-sticky-module
- added nginx-sticky-module-ng version 1.2.5</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-3" date="1432209600">- added support to systemd (RHEL/CentOS &gt;= 7)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.0-1" date="1442923200">- update to 1.9.0 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.1-1" date="1442923201">- update to 1.9.1 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.2-1" date="1442923202">- update to 1.9.2 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.3-1" date="1442923203">- update to 1.9.3 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.4-1" date="1442923204">- update to 1.9.4 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.5-1" date="1442923205">- update to 1.9.5 (mainline version)
- removed obsolete spdy module
- added experimental http2 module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.6-1" date="1449835200">- update to 1.9.6 (mainline version)</changelog>
</package>
<package pkgid="b901dd6ae120c9618b913cee24fb4dec39bfcad169ecfa558dc50b910cf6a6f8" name="nginx" arch="src">
  <version epoch="0" ver="1.9.7" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-2" date="1430913600">- updated nginx-push-stream-module to version 0.5.1
- removed obsolete nginx-sticky-module
- added nginx-sticky-module-ng version 1.2.5</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-3" date="1432209600">- added support to systemd (RHEL/CentOS &gt;= 7)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.0-1" date="1442923200">- update to 1.9.0 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.1-1" date="1442923201">- update to 1.9.1 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.2-1" date="1442923202">- update to 1.9.2 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.3-1" date="1442923203">- update to 1.9.3 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.4-1" date="1442923204">- update to 1.9.4 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.5-1" date="1442923205">- update to 1.9.5 (mainline version)
- removed obsolete spdy module
- added experimental http2 module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.6-1" date="1449835200">- update to 1.9.6 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.7-1" date="1449835201">- update to 1.9.7 (mainline version)</changelog>
</package>
<package pkgid="54163b29bab4e47c37c42f1344884b9b92811ae2bda691e51504d5d9cbc7ac3e" name="nginx" arch="src">
  <version epoch="0" ver="1.9.8" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.8.0-3" date="1432209600">- added support to systemd (RHEL/CentOS &gt;= 7)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.0-1" date="1442923200">- update to 1.9.0 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.1-1" date="1442923201">- update to 1.9.1 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.2-1" date="1442923202">- update to 1.9.2 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.3-1" date="1442923203">- update to 1.9.3 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.4-1" date="1442923204">- update to 1.9.4 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.5-1" date="1442923205">- update to 1.9.5 (mainline version)
- removed obsolete spdy module
- added experimental http2 module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.6-1" date="1449835200">- update to 1.9.6 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.7-1" date="1449835201">- update to 1.9.7 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.8-1" date="1449835202">- update to 1.9.8 (mainline version)</changelog>
</package>
<package pkgid="f5f8ffddf5b111c941a3efc1dba6962fca31a0a5699a7bbd493314bda527b166" name="nginx" arch="src">
  <version epoch="0" ver="1.9.9" rel="1.el7_9"/>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.0-1" date="1442923200">- update to 1.9.0 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.1-1" date="1442923201">- update to 1.9.1 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.2-1" date="1442923202">- update to 1.9.2 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.3-1" date="1442923203">- update to 1.9.3 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.4-1" date="1442923204">- update to 1.9.4 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.5-1" date="1442923205">- update to 1.9.5 (mainline version)
- removed obsolete spdy module
- added experimental http2 module</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.6-1" date="1449835200">- update to 1.9.6 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.7-1" date="1449835201">- update to 1.9.7 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.8-1" date="1449835202">- update to 1.9.8 (mainline version)</changelog>
  <changelog author="Azevedo &lt;azevedo@voipdomain.io&gt; - 1.9.9-1" date="1449835203">- update to 1.9.9 (mainline version)</changelog>
</package>
<package pkgid="e88aa0d6b917ae2a5d3338e2356513b47142d2d477e88bb7dc6cca23624e2ab1" name="ngrep" arch="src">
  <version epoch="0" ver="1.45" rel="8.git20131221.16ba99a.el7_9"/>
  <changelog author="Patrick &quot;Jima&quot; Laughton &lt;jima@beer.tclug.org&gt; 1.45-2" date="1187697600">- Rebuild for BuildID</changelog>
  <changelog author="Patrick &quot;Jima&quot; Laughton &lt;jima@beer.tclug.org&gt; 1.45-3" date="1187784000">- License clarification</changelog>
  <changelog author="Patrick &quot;Jima&quot; Laughton &lt;jima@beer.tclug.org&gt; 1.45-4" date="1202904000">- Bump-n-build for GCC 4.3</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.45-5" date="1235563200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.45-6" date="1248523200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="Oliver Falk &lt;oliver@linux-kernel.at&gt; 1.45-6.1" date="1254312000">- Bump-n-build for new libpcap</changelog>
  <changelog author="Oliver Falk &lt;oliver@linux-kernel.at&gt; 1.45-6.2" date="1254312001">- Next shot with --with-pcap-includes (Thanks Moroslav!)</changelog>
  <changelog author="Oliver Falk &lt;oliver@linux-kernel.at&gt; 1.45-6.3" date="1268740800">- Rebuild to (hopefully) fix pcap bpf.h not beeing found</changelog>
  <changelog author="Oliver Falk &lt;oliver@linux-kernel.at&gt; 1.45-7" date="1268740801">- pcap has moved to /usr/include/pcap, instead of /usr/include :-/</changelog>
  <changelog author="Christopher Meng &lt;rpm@cicku.me&gt; - 1.45-8.git20131221.16ba99a" date="1387627200">- Checkout from official repo(BZ#1044630).
- Remove patch for system pcre as configure script can handle it now.
- Add format security check fix due to dumb GCC.</changelog>
</package>
<package pkgid="6a6c376b526429da12d798d6558b9e88aac411adba7290ab0341b662307dee8f" name="nkf" arch="src">
  <version epoch="1" ver="2.1.3" rel="5.el7_9"/>
  <changelog author="Akira TAGOH &lt;tagoh@redhat.com&gt; - 1:2.1.2-5" date="1353585600">- the spec file cleanup.</changelog>
  <changelog author="Akira TAGOH &lt;tagoh@redhat.com&gt; - 1:2.1.2-6" date="1353672000">- Add %check.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:2.1.2-7" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1:2.1.2-8" date="1374062400">- Perl 5.18 rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:2.1.2-9" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Akira TAGOH &lt;tagoh@redhat.com&gt; - 1:2.1.3-1" date="1385467200">- New upstream release. (#1034089)</changelog>
  <changelog author="Akira TAGOH &lt;tagoh@redhat.com&gt; - 1:2.1.3-2" date="1389787200">- Fix the broken nkf man page. (#1039359)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:2.1.3-3" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:2.1.3-4" date="1408276800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 1:2.1.3-5" date="1409054400">- Perl 5.20 rebuild</changelog>
</package>
<package pkgid="06af1759206314a42ba95f1d447f0090a9f0477953774b85c5ee420d1c34477f" name="nodejs" arch="src">
  <version epoch="1" ver="6.17.1" rel="1.el7_9"/>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1:6.12.2-1" date="1512734400">- Update to 6.12.2
- https://nodejs.org/en/blog/release/v6.12.2/</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1:6.12.3-1" date="1515672000">- Update to 6.12.3
- https://nodejs.org/en/blog/release/v6.12.3/</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1:6.13.1-1" date="1520510400">- Update to 6.13.1
- https://nodejs.org/en/blog/release/v6.12.4/
- https://nodejs.org/en/blog/release/v6.12.5/
- https://nodejs.org/en/blog/release/v6.13.0/
- https://nodejs.org/en/blog/release/v6.13.1/</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1:6.14.0-1" date="1522238400">- Update to 6.14.0
- https://nodejs.org/en/blog/release/v6.14.0/</changelog>
  <changelog author="Rafael dos Santos &lt;rdossant@redhat.com&gt; - 1:6.14.0-2" date="1523620800">- Use standard Fedora linker flags (bug #1543859)</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1:6.14.2-1" date="1525348800">- Update to 6.14.2
- https://nodejs.org/en/blog/release/v6.14.2/
- https://nodejs.org/en/blog/release/v6.14.1/</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1:6.14.3-1" date="1528977600">- Update to 6.14.3 security release
- https://nodejs.org/en/blog/release/v6.14.3/</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1:6.16.0-1" date="1547121600">- Update to 6.16.0 security release
- https://nodejs.org/en/blog/release/v6.16.0/
- https://nodejs.org/en/blog/release/v6.15.1/
- https://nodejs.org/en/blog/release/v6.15.0/
- https://nodejs.org/en/blog/release/v6.14.4/</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1:6.16.0-2" date="1547121601">- Bundle http-parser since it uses backported features not available in RHEL</changelog>
  <changelog author="Stephen Gallagher &lt;sgallagh@redhat.com&gt; - 1:6.17.1-1" date="1560859200">- Update to 6.17.1 security release
- https://nodejs.org/en/blog/release/v6.17.1/
- https://nodejs.org/en/blog/release/v6.17.0/
- Drop debug build</changelog>
</package>
<package pkgid="c313ee854987e702e8dae657fae81c8d2b24a90700f683e49e2b841b2a9504b8" name="nodejs-packaging" arch="src">
  <version epoch="0" ver="7" rel="1.el7_9"/>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 1-1" date="1371124800">- initial package</changelog>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 2-1" date="1371816000">- move multiple version list to /usr/share/node
- bump nodejs Requires to 0.10.12
- add Requires: redhat-rpm-config</changelog>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 3-1" date="1373457600">- dependencies and engines can be lists or strings too
- handle unversioned dependencies on multiply versioned modules correctly
  (RHBZ#982798)
- restrict to compatible arches</changelog>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 4-1" date="1375099200">- handle cases where the symlink target exists gracefully</changelog>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 6-1" date="1400932800">- nodejs-fixdep: use real option parsing
- nodejs-fixdep: support modifying optionalDependencies and devDependencies
- req: support the caret operator
- nodejs-symlink-deps: add --force option
- nodejs-symlink-deps: add --build alias for --check
- nodejs-fixdep: support converting to caret dependencies
- nodejs-fixdep: support non-dictionary dependency properties
- multiver_modules: add nan</changelog>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 7-1" date="1401019200">- nodejs-symlink-deps: fix regression preventing multiply versioned modules from
  being symlinked correctly</changelog>
</package>
<package pkgid="c2564332ce4812c844efcded2551fdd082111713f65ec0f4da9172dcc826ac58" name="npm" arch="src">
  <version epoch="0" ver="1.3.6" rel="5.el7_9"/>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 1.2.17-3" date="1366027200">- rebuilt</changelog>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 1.2.17-4" date="1366027201">- add EPEL dependency generation macro</changelog>
  <changelog author="Stanislav Ochotnicky &lt;sochotnicky@redhat.com&gt; - 1.2.17-5" date="1366200000">- Fix manpage names so that npm help finds them</changelog>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 1.2.17-6" date="1371902400">- restrict to compatible arches</changelog>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 1.3.0-1" date="1371988800">- new upstream release 1.3.0
- use system paths for manual pages and documentation (RHBZ#953051)</changelog>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 1.3.3-1" date="1373630400">- new upstream release 1.3.3
- fixes insecure temporary directory generation (CVE-2013-4116; RHBZ#983917)</changelog>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 1.3.6-1" date="1375185600">- new upstream release 1.3.6</changelog>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 1.3.6-2" date="1375185601">- license changed from MITNFA to Artistic 2.0</changelog>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 1.3.6-4" date="1375963200">- remove unnecessary symlink to mandir
  fixes upgrade path from certain older versions of npm</changelog>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 1.3.6-5" date="1397476800">- backport fix for traceback encountered when using "npm search" (RHBZ#1087065)</changelog>
</package>
<package pkgid="c820b80feca51c752de3fe2d0ae69fc8a25d16c3965eacf847afbde0472bada6" name="ntfs-3g" arch="src">
  <version epoch="2" ver="2017.3.23" rel="6.el7_9"/>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2:2016.2.22-3" date="1478088000">- enable posix ACLS
- enable xattr mappings</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2:2016.2.22-4" date="1486555200">- apply patch for CVE-2017-0358
- NOTE: Fedora does not setuid ntfs-3g, so it should not be vulnerable 
  but some users might make this change so we applied the patch anyways</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2:2017.3.23-1" date="1496145600">- update to 2017.3.23</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2:2017.3.23-2" date="1501156800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2:2017.3.23-3" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2:2017.3.23-3.1" date="1513771200">- test build with patch from Jean-Pierre André to fix the $MFT/$MFTMirr mismatch</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2:2017.3.23-4" date="1515067200">- use upstream tarball again (non-free file is removed)
- remove unused CVE-2015-3202 patch</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2:2017.3.23-5" date="1518091200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt;" date="1525694400">- big sectors patch from Jean-Pierre André</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2:2017.3.23-6" date="1526904000">- apply updated big sectors patch</changelog>
</package>
<package pkgid="d00536b0a8ed5f91b85dba683a819005ec4129f23fc926a41d1e2d2ed38d3f5d" name="nv-codec-headers" arch="src">
  <version epoch="0" ver="8.1.24.11" rel="1.el7_9"/>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 8.0.14.1-1" date="1519732800">- First build</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 8.1.24.1-1" date="1523793600">- Update to 8.1.24.1</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 8.1.24.2-1" date="1528891200">- Update to 8.1.24.2</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 8.1.24.2-2" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 8.1.24.8-1" date="1549195200">- Update to 8.1.24.8</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 8.1.24.9-1" date="1557576000">- Update to 8.1.24.9</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 8.1.24.9-2" date="1558440000">- Use correct path for pkg-config file</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 8.1.24.10-1" date="1569326400">- Update to 8.1.24.10</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@gmail.com&gt; - 8.1.24.11-1" date="1576497600">- Update to 8.1.24.11</changelog>
</package>
<package pkgid="53b1c921b59480f5860ba6a2c2b60b47fefb4b8bd335f940025717f1dc5f8d21" name="nv-codec-headers" arch="src">
  <version epoch="0" ver="9.1.23.1" rel="1.el7_9"/>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 8.2.15.5-1" date="1541505600">- Update to 8.2.15.5</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 8.2.15.6-1" date="1546689600">- Update to 8.2.15.6</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 8.2.15.6-2" date="1549022400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 8.2.15.7-1" date="1549195200">- Update to 8.2.15.7</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 9.0.18.0-1" date="1551441600">- Update to 9.0.18.0</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 9.0.18.1-1" date="1557576000">- Update to 9.0.18.1</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 9.0.18.1-2" date="1558440000">- Use correct path for pkg-config file</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 9.0.18.1-3" date="1564056000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 9.1.23.0-1" date="1569326400">- Update to 9.1.23.0</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@gmail.com&gt; - 9.1.23.1-1" date="1576497600">- Update to 9.1.23.1</changelog>
</package>
<package pkgid="0346abade854c4d74b850782770a3545ef2b5d9b8d4b5cf2bfe00ce1ac0fbfee" name="nvenc" arch="src">
  <version epoch="0" ver="5.0.1" rel="1.el7_9"/>
  <changelog author="Simone Caronni &lt;negativo17@gmail.com&gt;" date="1428667200">- First build.</changelog>
</package>
<package pkgid="1f7d650b0f5b9a32d6aa01a3dcd5b4501a815f9137b2e2571cb8a767b85c061f" name="ocl-icd" arch="src">
  <version epoch="0" ver="2.2.12" rel="1.el7_9"/>
  <changelog author="Fran&#xE7;ois Cami &lt;fcami@fedoraproject.org&gt; - 2.2.8-1.git20151217.0122332" date="1450699200">- Update to 2.2.8.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.2.8-2.git20151217.0122332" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;fedora@besser82.io&gt; - 2.2.8-3.git20151217.0122332" date="1460116800">- add Requires for virtual Provides: opencl-icd (RHBZ #1317600)
- add rubygems and rubypick to BuildRequires</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenko@redhat.com&gt; - 2.2.9-1" date="1471176000">- Update to 2.2.9
- Drop requires for opencl-icd</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenko@redhat.com&gt; - 2.2.9-2" date="1472644800">- Rebuild for OpenCL 2.1</changelog>
  <changelog author="Orion Poplawski &lt;orion@cora.nwra.com&gt; - 2.2.9-3" date="1479729600">- Drop unneeded BR on rubypick</changelog>
  <changelog author="Igor Gnatenko &lt;i.gnatenko.brain@gmail.com&gt; - 2.2.10-1" date="1480852800">- Update to 2.2.10</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenko@redhat.com&gt; - 2.2.11-1" date="1484913600">- Update to 2.2.11 (RHBZ #1415150)</changelog>
  <changelog author="Simone Caronni &lt;negativo17@gmail.com&gt; - 2.2.11-2" date="1489665600">- Allow building on CentOS/RHEL 6.</changelog>
  <changelog author="Simone Caronni &lt;negativo17@gmail.com&gt; - 2.2.12-1" date="1521806400">- Update to 2.2.12, adds OpenCL 2.2 support.</changelog>
</package>
<package pkgid="85fe5804ad98bd502951de3405685991c41a2ffbef1fb2cc742400c33d297c35" name="oniguruma" arch="src">
  <version epoch="0" ver="6.8.2" rel="1.el7_9"/>
  <changelog author="Mamoru Tasaka &lt;mtasaka@fedoraproject.org&gt; - 5.9.2-3" date="1325764800">- F-17: rebuild against gcc47</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 5.9.2-4" date="1342785600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 5.9.3-1" date="1357300800">- 5.9.3</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 5.9.3-2" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 5.9.4-1" date="1367236800">- 5.9.4</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 5.9.4-2" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 5.9.5-1" date="1384171200">- 5.9.5</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 5.9.5-2" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 5.9.5-3" date="1408276800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Carl George &lt;carl@george.computer&gt; - 6.8.2-1" date="1590494400">- Rebase to 6.8.2 rhbz#1777660</changelog>
</package>
<package pkgid="a586333f7dbbb1b6bfa0cbafe67b2197b49365c1cf16333d7895b2c49f3827fb" name="openal-soft" arch="src">
  <version epoch="0" ver="1.16.0" rel="3.el7_9"/>
  <changelog author="Dan Hor&#xE1;k &lt;dan[at]danny.cz&gt; - 1.14-2" date="1334232000">- the used fpu control bits are x86 specific</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.14-3" date="1342785600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fran&#xE7;ois Cami &lt;fcami@fedoraproject.org&gt; - 1.15.1-1" date="1357905600">- New upstream release</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.15.1-2" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.15.1-3" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.15.1-4" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.15.1-5" date="1408276800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Fran&#xE7;ois Cami &lt;fcami@fedoraproject.org&gt; - 1.16.0-1" date="1408276801">- New upstream release</changelog>
  <changelog author="Fran&#xE7;ois Cami &lt;fcami@fedoraproject.org&gt; - 1.16.0-2" date="1409054400">- Add the -qt subpackage to host the alsoft-config tool</changelog>
  <changelog author="Marcin Juszkiewicz &lt;mjuszkiewicz@redhat.com&gt; - 1.16.0-3" date="1410177600">- Check for arm_neon.h only on 32bit ARM</changelog>
</package>
<package pkgid="333c59faff4bcac40e3ec01fccbfa40bddfacab7cc72fa49b0734cdb4eeb7ee5" name="opencl-headers" arch="src">
  <version epoch="0" ver="2.2" rel="1.20180306gite986688.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.2-3" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fabian Deutsch &lt;fabiand@fedoraproject.org&gt; - 1.2-4" date="1398427200">- Add patch for cl.hpp to be usable on arm rhbz#1027199</changelog>
  <changelog author="Fabian Deutsch &lt;fabiand@fedoraproject.org&gt; - 1.2-5" date="1398427201">- Pull patch application into pre</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.2-6" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.2-7" date="1434542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Dave Airlie &lt;airlied@redhat.com&gt; - 1.2-8" date="1453032000">- add cl_egl.h</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2-9" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenko@redhat.com&gt; - 2.1-1" date="1472644800">- Update to 2.1</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.1-2" date="1486814400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Simone Caronni &lt;negativo17@gmail.com&gt; - 2.2-1.20180306gite986688" date="1521806400">- Update to 2.2.
- Use packaging guidelines for snapshots.</changelog>
</package>
<package pkgid="d205d6182b546bc29d4145c3ea460c607a7a1481d00c9c534a6b1c9446f6bd26" name="opencore-amr" arch="src">
  <version epoch="0" ver="0.1.3" rel="3.el7_9"/>
  <changelog author="Hans de Goede &lt;j.w.r.degoede@hhs.nl&gt; 0.1.1-1" date="1248955200">- First version of the RPM Fusion package</changelog>
  <changelog author="Hans de Goede &lt;j.w.r.degoede@hhs.nl&gt; 0.1.2-1" date="1254657600">- New upstream release 0.1.2</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.2-2" date="1327492800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.2-3" date="1330689600">- Rebuilt for c++ ABI breakage</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.3-1" date="1336910400">- Update to 0.1.3</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.3-2" date="1337342400">- Fix pkgconfig include</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.3-3" date="1362312000">- Mass rebuilt for Fedora 19 Features</changelog>
</package>
<package pkgid="d0f09f2aea7323287ba62183856dc48b7cf83df409d8127b33308289ea8036e5" name="opencore-amr" arch="src">
  <version epoch="0" ver="0.1.5" rel="6.el7_9"/>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.3-1" date="1336910400">- Update to 0.1.3</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.3-2" date="1337342400">- Fix pkgconfig include</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.3-3" date="1362312000">- Mass rebuilt for Fedora 19 Features</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.1.3-4" date="1409486400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 0.1.5-1" date="1490011200">- Update to 0.1.5</changelog>
  <changelog author="RPM Fusion Release Engineering &lt;kwizart@rpmfusion.org&gt; - 0.1.5-2" date="1504180800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="RPM Fusion Release Engineering &lt;leigh123linux@googlemail.com&gt; - 0.1.5-3" date="1519905600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="RPM Fusion Release Engineering &lt;leigh123linux@gmail.com&gt; - 0.1.5-4" date="1532692800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 0.1.5-5" date="1534680000">- Rebuilt for Fedora 29 Mass Rebuild binutils issue</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.5-6" date="1544529600">- Spec file clean-up</changelog>
</package>
<package pkgid="3e93715068b64b0d2a80b2d3065e18983be690b3ba3c5d7a2a777beb96e64cb6" name="openjpeg2" arch="src">
  <version epoch="0" ver="2.3.0" rel="6.el7_9"/>
  <changelog author="Sandro Mani &lt;manisandro@gmail.com&gt; - 2.2.0-1" date="1502366400">- Update to 2.2.0</changelog>
  <changelog author="Sandro Mani &lt;manisandro@gmail.com&gt; - 2.2.0-2" date="1504180800">- Backport patch for CVE-2017-12982</changelog>
  <changelog author="Sandro Mani &lt;manisandro@gmail.com&gt; - 2.2.0-3" date="1504180801">- Backport more security fixes, including for CVE-2017-14041 and CVE-2017-14040</changelog>
  <changelog author="Sandro Mani &lt;manisandro@gmail.com&gt; - 2.2.0-4" date="1504785600">- Backport fix for CVE-2017-14039</changelog>
  <changelog author="Sandro Mani &lt;manisandro@gmail.com&gt; - 2.3.0-1" date="1507204800">- Update to 2.3.0</changelog>
  <changelog author="Sandro Mani &lt;manisandro@gmail.com&gt; - 2.3.0-2" date="1514203200">- Use BUILD_STATIC_LIBS=OFF instead of deleting the static library after build</changelog>
  <changelog author="Sandro Mani &lt;manisandro@gmail.com&gt; - 2.3.0-3" date="1514203201">- Rename tool names at cmake level to ensure OpenJPEGTargets.cmake refers to the renamed files</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 2.3.0-4" date="1517659200">- Switch to %ldconfig_scriptlets</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 2.3.0-5" date="1518091200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Sandro Mani &lt;manisandro@gmail.com&gt; - 2.3.0-6" date="1519041600">- Add missing BR: gcc, make</changelog>
</package>
<package pkgid="4da21450601852af73414b2a9dd973dd4c71a4d96402985b0820789e4b136e62" name="openpgm" arch="src">
  <version epoch="0" ver="5.2.122" rel="2.el7_9"/>
  <changelog author="Jose Pedro Oliveira &lt;jpo at di.uminho.pt&gt; - 5.1.118-0" date="1355832000">- First Fedora specfile</changelog>
  <changelog author="Jose Pedro Oliveira &lt;jpo at di.uminho.pt&gt; - 5.1.118-1" date="1355918400">- Change license from LGPLv2.1 to LGPLv2 (867182#c13)</changelog>
  <changelog author="Jose Pedro Oliveira &lt;jpo at di.uminho.pt&gt; - 5.1.118-2" date="1356091200">- Renamed the tarball (replaced '%7E' by '~')
- Removed the defattr lines</changelog>
  <changelog author="Jose Pedro Oliveira &lt;jpo at di.uminho.pt&gt; - 5.1.118-3" date="1356091201">- Build requires python (no longer available by default in F18+ buildroots)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 5.1.118-4" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Jose Pedro Oliveira &lt;jpo at di.uminho.pt&gt; - 5.2.122-1" date="1368619200">- Update to 5.2.122</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 5.2.122-2" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
</package>
<package pkgid="188a17ae637c5ad9b84d36c0d783f42846c6a8541effdc4f69a542bff29eae8c" name="openr2" arch="src">
  <version epoch="0" ver="1.3.3" rel="1.el7_9"/>
  <changelog author="dlucio &lt;dlucio&gt; 1.3.2-1.mga2" date="1332763200">+ Revision: 226491
- imported package openr2</changelog>
  <changelog author="umeabot &lt;umeabot&gt; 1.3.2-2.mga3" date="1358078400">+ Revision: 362108
- Mass Rebuild - https://wiki.mageia.org/en/Feature:Mageia3MassRebuild</changelog>
  <changelog author="umeabot &lt;umeabot&gt; 1.3.2-3.mga4" date="1382097600">+ Revision: 508700
- Mageia 4 Mass Rebuild</changelog>
  <changelog author="umeabot &lt;umeabot&gt; 1.3.2-4.mga5" date="1410868800">+ Revision: 683230
- Mageia 5 Mass Rebuild</changelog>
  <changelog author="umeabot &lt;umeabot&gt; 1.3.2-5.mga5" date="1413374400">+ Revision: 743488
- Second Mageia 5 Mass Rebuild</changelog>
  <changelog author="oden &lt;oden&gt; 1.3.3-1.mga6" date="1447675200">+ Revision: 903654
- 1.3.3
- libify properly but waste no time upgrading cleanly</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.3-2" date="1540468800">- Revised to CentOS build</changelog>
</package>
<package pkgid="c5f28c5460ff67c964c5f8a80dd2a542535e4ac2828adabdf9d00c3189b5cfe6" name="opensips" arch="src">
  <version epoch="0" ver="2.2.0" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.1-2" date="1345204800">- Enabled json module
- Enabled xmlrpc module
- Enabled cachedb_memcached module on EL5, EL6
- Enabled cachedb_redis module on EL6</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 1.8.1-3" date="1348315200">- rebuild against libmemcached.so.11 without SASL</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-1" date="1352203200">- Ver. 1.8.2 (Bugfix release)</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-2" date="1357819200">- Allow rtpproxy module to accept avps
- Few bugfixes</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
</package>
<package pkgid="68fb8400b7307587c73c5a683eb65ef5cd9e4452e84c44e5ecb02770034e1a2a" name="opensips" arch="src">
  <version epoch="0" ver="2.2.1" rel="1.el7_9"/>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 1.8.1-3" date="1348315200">- rebuild against libmemcached.so.11 without SASL</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-1" date="1352203200">- Ver. 1.8.2 (Bugfix release)</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-2" date="1357819200">- Allow rtpproxy module to accept avps
- Few bugfixes</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1476446400">- Update to 2.2.1</changelog>
</package>
<package pkgid="f7e93e451ea45336c0e280683a2fb37656db054b25fb6a50b21f5b6c275da604" name="opensips" arch="src">
  <version epoch="0" ver="2.2.2" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-1" date="1352203200">- Ver. 1.8.2 (Bugfix release)</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-2" date="1357819200">- Allow rtpproxy module to accept avps
- Few bugfixes</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1476446400">- Update to 2.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.2-1" date="1477915200">- Update to 2.2.2</changelog>
</package>
<package pkgid="cfa8603010eb1b6ce780a8649d0b1ead18740310cc9629197553b9cded19f8c6" name="opensips" arch="src">
  <version epoch="0" ver="2.2.3" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-2" date="1357819200">- Allow rtpproxy module to accept avps
- Few bugfixes</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1476446400">- Update to 2.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.2-1" date="1477915200">- Update to 2.2.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.3-1" date="1554206400">- Update to 2.2.3</changelog>
</package>
<package pkgid="91e6bc075f0bf07b0b8a2fce80a63b19fe79a953005f295665cbeb7c46bbb677" name="opensips" arch="src">
  <version epoch="0" ver="2.2.4" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1476446400">- Update to 2.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.2-1" date="1477915200">- Update to 2.2.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.3-1" date="1554206400">- Update to 2.2.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.4-1" date="1554206401">- Update to 2.2.4</changelog>
</package>
<package pkgid="8460f9cfce0b9afcc534019d6c99a65e59fb3b4c2b99c7d547e166dda7e5641f" name="opensips" arch="src">
  <version epoch="0" ver="2.2.5" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1476446400">- Update to 2.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.2-1" date="1477915200">- Update to 2.2.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.3-1" date="1554206400">- Update to 2.2.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.4-1" date="1554206401">- Update to 2.2.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.5-1" date="1554206402">- Update to 2.2.5</changelog>
</package>
<package pkgid="ae76bfbf3bff64cd96b3e0b8dcda6bcb6c853143422d2b648234c84b7ef98402" name="opensips" arch="src">
  <version epoch="0" ver="2.2.6" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1476446400">- Update to 2.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.2-1" date="1477915200">- Update to 2.2.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.3-1" date="1554206400">- Update to 2.2.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.4-1" date="1554206401">- Update to 2.2.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.5-1" date="1554206402">- Update to 2.2.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.6-1" date="1554206403">- Update to 2.2.6</changelog>
</package>
<package pkgid="022571a08fc3a7c479a73a2cd07ada7180a550377c0823dd946b31d297e8ccde" name="opensips" arch="src">
  <version epoch="0" ver="2.2.7" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.1-1" date="1476446400">- Update to 2.2.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.2-1" date="1477915200">- Update to 2.2.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.3-1" date="1554206400">- Update to 2.2.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.4-1" date="1554206401">- Update to 2.2.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.5-1" date="1554206402">- Update to 2.2.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.6-1" date="1554206403">- Update to 2.2.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.2.7-1" date="1554206404">- Update to 2.2.7</changelog>
</package>
<package pkgid="c3d320cd09fc61adb66ff093fb9348d968e7f8d8e429d7f779e2bb8fd5f9e0e2" name="opensips" arch="src">
  <version epoch="0" ver="2.3.0" rel="1.el7_9"/>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 1.8.1-3" date="1348315200">- rebuild against libmemcached.so.11 without SASL</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-1" date="1352203200">- Ver. 1.8.2 (Bugfix release)</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-2" date="1357819200">- Allow rtpproxy module to accept avps
- Few bugfixes</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
</package>
<package pkgid="5e48c43164a29289e6fc51c1a91aa2afe0320921c009e8fda3be93acd9ef3b18" name="opensips" arch="src">
  <version epoch="0" ver="2.3.1" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-1" date="1352203200">- Ver. 1.8.2 (Bugfix release)</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-2" date="1357819200">- Allow rtpproxy module to accept avps
- Few bugfixes</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1554206400">- Update to 2.3.1</changelog>
</package>
<package pkgid="9ff495067645d00526c837091853164d489d6b578d2b6b7702e938304373cf7e" name="opensips" arch="src">
  <version epoch="0" ver="2.3.2" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-2" date="1357819200">- Allow rtpproxy module to accept avps
- Few bugfixes</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1554206400">- Update to 2.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.2-1" date="1554206401">- Update to 2.3.2</changelog>
</package>
<package pkgid="f82cd5278550c95d98ed6f8f9d264e6d3283de815e2f9d4d73ffbfa82a4c8b1e" name="opensips" arch="src">
  <version epoch="0" ver="2.3.3" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1554206400">- Update to 2.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.2-1" date="1554206401">- Update to 2.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.3-1" date="1554206402">- Update to 2.3.3</changelog>
</package>
<package pkgid="df584382e51fdffa4c0d30f7c209d0a25bfe34a7d43ff54cebc7e447289837a0" name="opensips" arch="src">
  <version epoch="0" ver="2.3.4" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1554206400">- Update to 2.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.2-1" date="1554206401">- Update to 2.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.3-1" date="1554206402">- Update to 2.3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.4-1" date="1554206403">- Update to 2.3.4</changelog>
</package>
<package pkgid="3aa9c361dc5c882a9970f11037c1911f521015d0e2aff5cb53307f19698513ea" name="opensips" arch="src">
  <version epoch="0" ver="2.3.5" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1554206400">- Update to 2.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.2-1" date="1554206401">- Update to 2.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.3-1" date="1554206402">- Update to 2.3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.4-1" date="1554206403">- Update to 2.3.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.5-1" date="1554206404">- Update to 2.3.5</changelog>
</package>
<package pkgid="7455ed83483c2555f2cacfe4e92067cf1f57bb435307f5debbd310de38c735b0" name="opensips" arch="src">
  <version epoch="0" ver="2.3.6" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.1-1" date="1554206400">- Update to 2.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.2-1" date="1554206401">- Update to 2.3.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.3-1" date="1554206402">- Update to 2.3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.4-1" date="1554206403">- Update to 2.3.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.5-1" date="1554206404">- Update to 2.3.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.3.6-1" date="1554206405">- Update to 2.3.6</changelog>
</package>
<package pkgid="3b71a7cb9f9adcb37ccc61b32352cf2972e3dea442ba9c0d7f46cd5b18b52b24" name="opensips" arch="src">
  <version epoch="0" ver="2.4.0" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-1" date="1352203200">- Ver. 1.8.2 (Bugfix release)</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-2" date="1357819200">- Allow rtpproxy module to accept avps
- Few bugfixes</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
</package>
<package pkgid="873ee764e0e942e1d39c1f4691b35e926dc3fd10fbf1a2d2bf9d54609b4c1778" name="opensips" arch="src">
  <version epoch="0" ver="2.4.1" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-2" date="1357819200">- Allow rtpproxy module to accept avps
- Few bugfixes</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.1-1" date="1554206400">- Update to 2.4.1</changelog>
</package>
<package pkgid="1e3fb3d67b49145f3d874063d3456879b764ada630e4976570c5c3cf072a15ff" name="opensips" arch="src">
  <version epoch="0" ver="2.4.10" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.1-1" date="1554206400">- Update to 2.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.2-1" date="1554206401">- Update to 2.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.3-1" date="1554206402">- Update to 2.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.4-1" date="1554206403">- Update to 2.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.5-1" date="1554206404">- Update to 2.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.6-1" date="1560254400">- Update to 2.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.7-1" date="1580212800">- Update to 2.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.8-1" date="1593518400">- Update to 2.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.9-1" date="1605614400">- Update to 2.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.10-1" date="1620302400">- Update to 2.4.10</changelog>
</package>
<package pkgid="9e8b21ac2f5b69a23bc7bd47efff92a787f85bf7c2a07b369c497855f7a9a03e" name="opensips" arch="src">
  <version epoch="0" ver="2.4.11" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.2-1" date="1554206400">- Update to 2.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.3-1" date="1554206401">- Update to 2.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.4-1" date="1554206402">- Update to 2.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.5-1" date="1554206403">- Update to 2.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.6-1" date="1560254400">- Update to 2.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.7-1" date="1580212800">- Update to 2.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.8-1" date="1593518400">- Update to 2.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.9-1" date="1605614400">- Update to 2.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.10-1" date="1620302400">- Update to 2.4.10</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.11-1" date="1624363200">- Update to 2.4.11</changelog>
</package>
<package pkgid="9b09c73e4f60e03d2d335d1432f791cfe8afbde66f3517c2bd0d711eb46f3e8f" name="opensips" arch="src">
  <version epoch="0" ver="2.4.2" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.1-1" date="1554206400">- Update to 2.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.2-1" date="1554206401">- Update to 2.4.2</changelog>
</package>
<package pkgid="f3291e8d61263bccd101c927a744b42385e8c1c05599dbed51b3cabcdfa21405" name="opensips" arch="src">
  <version epoch="0" ver="2.4.3" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.1-1" date="1554206400">- Update to 2.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.2-1" date="1554206401">- Update to 2.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.3-1" date="1554206402">- Update to 2.4.3</changelog>
</package>
<package pkgid="60b64d55ea3146fe587b3beaf89890e58f03f04d595d1b37c67f3e2594043f88" name="opensips" arch="src">
  <version epoch="0" ver="2.4.4" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.1-1" date="1554206400">- Update to 2.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.2-1" date="1554206401">- Update to 2.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.3-1" date="1554206402">- Update to 2.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.4-1" date="1554206403">- Update to 2.4.4</changelog>
</package>
<package pkgid="7ed85f0bd86c227433228d9f0045d3aa777c469fa4a42185940f335511f8e88b" name="opensips" arch="src">
  <version epoch="0" ver="2.4.5" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.1-1" date="1554206400">- Update to 2.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.2-1" date="1554206401">- Update to 2.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.3-1" date="1554206402">- Update to 2.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.4-1" date="1554206403">- Update to 2.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.5-1" date="1554206404">- Update to 2.4.5</changelog>
</package>
<package pkgid="0b70f24b7c3189953c93473ab4256bfed67323e9f6148928619dad7ce8d0b066" name="opensips" arch="src">
  <version epoch="0" ver="2.4.6" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.1-1" date="1554206400">- Update to 2.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.2-1" date="1554206401">- Update to 2.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.3-1" date="1554206402">- Update to 2.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.4-1" date="1554206403">- Update to 2.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.5-1" date="1554206404">- Update to 2.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.6-1" date="1560254400">- Update to 2.4.6</changelog>
</package>
<package pkgid="83c393677967e6093fa21abeb5d372bdbd9b8f9e34b1f4f3fb62244242a9cb4f" name="opensips" arch="src">
  <version epoch="0" ver="2.4.7" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.1-1" date="1554206400">- Update to 2.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.2-1" date="1554206401">- Update to 2.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.3-1" date="1554206402">- Update to 2.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.4-1" date="1554206403">- Update to 2.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.5-1" date="1554206404">- Update to 2.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.6-1" date="1560254400">- Update to 2.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.7-1" date="1580212800">- Update to 2.4.7</changelog>
</package>
<package pkgid="37ea95479d279808ed5115970b79831e443557ee0a630e814c6f2bbe0ed7c53f" name="opensips" arch="src">
  <version epoch="0" ver="2.4.8" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.1-1" date="1554206400">- Update to 2.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.2-1" date="1554206401">- Update to 2.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.3-1" date="1554206402">- Update to 2.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.4-1" date="1554206403">- Update to 2.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.5-1" date="1554206404">- Update to 2.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.6-1" date="1560254400">- Update to 2.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.7-1" date="1580212800">- Update to 2.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.8-1" date="1593518400">- Update to 2.4.8</changelog>
</package>
<package pkgid="15ec1eff86f2d8f8c936650f421a89299bd3121f9945a65fd3ff75d3bfdd6eac" name="opensips" arch="src">
  <version epoch="0" ver="2.4.9" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.1-1" date="1554206400">- Update to 2.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.2-1" date="1554206401">- Update to 2.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.3-1" date="1554206402">- Update to 2.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.4-1" date="1554206403">- Update to 2.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.5-1" date="1554206404">- Update to 2.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.6-1" date="1560254400">- Update to 2.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.7-1" date="1580212800">- Update to 2.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.8-1" date="1593518400">- Update to 2.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.9-1" date="1605614400">- Update to 2.4.9</changelog>
</package>
<package pkgid="d0c91a6ddceea6a141c37f1e824a7119b6bd9788c12820244e1683936ed6fb4a" name="opensips" arch="src">
  <version epoch="0" ver="3.1.0" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="adbe4b7859391f2a9e3c3864a5facaae2703e00948bfafc202344097ec486650" name="opensips" arch="src">
  <version epoch="0" ver="3.1.1" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="cff12d2a2aa8f792cd4e86b572f208cc1b60356812739a5b4aa0082c5eba4b9c" name="opensips" arch="src">
  <version epoch="0" ver="3.1.10" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="835e85f5b355c2063ee5709fe8a2bcd961d45c4378b66420b96ab7719eddec0d" name="opensips" arch="src">
  <version epoch="0" ver="3.1.11" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="4be9f709c2adde78d1af9d575d87ff635c88bf3100853b1e2975425ea18fcf90" name="opensips" arch="src">
  <version epoch="0" ver="3.1.12" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="a962f8dd860b21218d600a7036fe62dc9fc899cb17a1eebb51ad47702819f0ae" name="opensips" arch="src">
  <version epoch="0" ver="3.1.13" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="ad353b0a9e00f0494436992a5c548ebf2e0b5747c5767d3011f473ff124016cd" name="opensips" arch="src">
  <version epoch="0" ver="3.1.14" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="9a333f7a4d2542fb90655dc75eac521bbc566315133911c1d4ed95bf0aae7618" name="opensips" arch="src">
  <version epoch="0" ver="3.1.15" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="96cc7dde16ad862cb6a72d7c4b19dc7a5c2fb76308a00505d6c4c3e54bb4ecec" name="opensips" arch="src">
  <version epoch="0" ver="3.1.16" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.1.16-1" date="1687348800">- OpenSIPS minor stable release: 3.1.16-1</changelog>
</package>
<package pkgid="a0c7470de1fadd275b8f0038a229a4a1b5a42773d0dc0955c5d1c57c7748436e" name="opensips" arch="src">
  <version epoch="0" ver="3.1.17" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.1.16-1" date="1687348800">- OpenSIPS minor stable release: 3.1.16-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.1.17-1" date="1693483200">- OpenSIPS minor stable release: 3.1.17-1</changelog>
</package>
<package pkgid="aa6160a7b4c45d7fc3f273dc84547a16233156687ded51cc85a62328ded68788" name="opensips" arch="src">
  <version epoch="0" ver="3.1.2" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="7bda27d911b9c01ff4ba24f8ff3f74cb3e5c35d79ca94f3b809967b5d6a61b8b" name="opensips" arch="src">
  <version epoch="0" ver="3.1.3" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="c737807e15ae9e715296b1ffe3ec52fe3753d3c65c0103afb450a785774a7446" name="opensips" arch="src">
  <version epoch="0" ver="3.1.4" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="4f4fa687bab550b5600cc21c17581c63a75d0ef1211e4d58d0e00470128e69c2" name="opensips" arch="src">
  <version epoch="0" ver="3.1.5" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="eb3452bad83f90474a53ad0a254ba3d890187b8e56259432d98e9a45db682886" name="opensips" arch="src">
  <version epoch="0" ver="3.1.6" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="c92367cfecf833318c9b00e12f84995e1b210446d56a432221ee5df8f97ba1a9" name="opensips" arch="src">
  <version epoch="0" ver="3.1.7" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="da756c5adf236d4cf71208f354a95ee444040e7fad9709c0dfd47e0d55537e77" name="opensips" arch="src">
  <version epoch="0" ver="3.1.8" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="98c35f8484c9c2425137700271047b817237b552d862e589bd97b81773c23a61" name="opensips" arch="src">
  <version epoch="0" ver="3.1.9" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 1.8.2-3" date="1358856000">- Revert systemd macros</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.9.1-1" date="1369224000">- Rebuild specification, add new modules and dependencies</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: callops, media_exchange, presence_dfks, qrouting,
  rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
</package>
<package pkgid="20bc43423e19aec604bdf0a3864566b610bae667353a1480a051858306cf83ee" name="opensips" arch="src">
  <version epoch="0" ver="3.2.0" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="15f3cdc2276d7bf37081b24840b8f5c6cd14059075f34442addf5a99b975b05b" name="opensips" arch="src">
  <version epoch="0" ver="3.2.10" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="ea37149006372954a619e6221ca72ca6bef14bdaaf4f70e64586b3d9428dad77" name="opensips" arch="src">
  <version epoch="0" ver="3.2.11" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="9a2a0655c6cfef8efcd7b09a8f32d69a2b60f4e10fb440d7379601a1949394b8" name="opensips" arch="src">
  <version epoch="0" ver="3.2.12" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="a43788b2bbd7aa2472c653bba7db08519a770b7aafde641ac0cd3cad44b5f8f1" name="opensips" arch="src">
  <version epoch="0" ver="3.2.13" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.13-1" date="1687348800">- OpenSIPS minor stable release: 3.2.13-1</changelog>
</package>
<package pkgid="589f799faa3178c71cd340c83e773b9bf5a2a122b4c7d6b44141d1506cb66a58" name="opensips" arch="src">
  <version epoch="0" ver="3.2.14" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.13-1" date="1687348800">- OpenSIPS minor stable release: 3.2.13-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.14-1" date="1693483200">- OpenSIPS minor stable release: 3.2.14-1</changelog>
</package>
<package pkgid="2cfec441f91d4eeeba9f2b29181918e7eaf3f9f618aaa3b5a3e94172266346bc" name="opensips" arch="src">
  <version epoch="0" ver="3.2.15" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.13-1" date="1687348800">- OpenSIPS minor stable release: 3.2.13-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.14-1" date="1693483200">- OpenSIPS minor stable release: 3.2.14-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.15-1" date="1697630400">- OpenSIPS minor stable release: 3.2.15-1</changelog>
</package>
<package pkgid="38253cd3cff2fd8abaf7a45b53576d3f247473b55d0ccb8abce24e00da63fa83" name="opensips" arch="src">
  <version epoch="0" ver="3.2.16" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.13-1" date="1687348800">- OpenSIPS minor stable release: 3.2.13-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.14-1" date="1693483200">- OpenSIPS minor stable release: 3.2.14-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.15-1" date="1697630400">- OpenSIPS minor stable release: 3.2.15-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.16-1" date="1703073600">- OpenSIPS minor stable release: 3.2.16-1</changelog>
</package>
<package pkgid="945e5f3d9144829b770b670129799a9198a59d46676c3f4a93249681cadded0d" name="opensips" arch="src">
  <version epoch="0" ver="3.2.18" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.13-1" date="1687348800">- OpenSIPS minor stable release: 3.2.13-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.14-1" date="1693483200">- OpenSIPS minor stable release: 3.2.14-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.15-1" date="1697630400">- OpenSIPS minor stable release: 3.2.15-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.16-1" date="1703073600">- OpenSIPS minor stable release: 3.2.16-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.17-1" date="1708516800">- OpenSIPS minor stable release: 3.2.17-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.18-1" date="1713441600">- OpenSIPS minor stable release: 3.2.18-1</changelog>
</package>
<package pkgid="b2029f31fed6a2a7fa3a634138545bea503fd353f4c8669ab694bbb34ff4f83e" name="opensips" arch="src">
  <version epoch="0" ver="3.2.19" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.13-1" date="1687348800">- OpenSIPS minor stable release: 3.2.13-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.14-1" date="1693483200">- OpenSIPS minor stable release: 3.2.14-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.15-1" date="1697630400">- OpenSIPS minor stable release: 3.2.15-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.16-1" date="1703073600">- OpenSIPS minor stable release: 3.2.16-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.17-1" date="1708516800">- OpenSIPS minor stable release: 3.2.17-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.18-1" date="1713441600">- OpenSIPS minor stable release: 3.2.18-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.2.19-1" date="1718798400">- OpenSIPS minor stable release: 3.2.19-1</changelog>
</package>
<package pkgid="ecd7144d71b6e0daaae208a0e188c59b2a15f78e5e3085f44f165e65484fb44b" name="opensips" arch="src">
  <version epoch="0" ver="3.2.2" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="7059a4cfff45c74b3370578ca83725fbea4c6ee8eaef279930b56dbba9514517" name="opensips" arch="src">
  <version epoch="0" ver="3.2.3" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="127805ebddbc05558ec36ea4cd84d9a61ac8e4a1dffeeedd6624198ba25472b4" name="opensips" arch="src">
  <version epoch="0" ver="3.2.4" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="5e9d803b49f54d06024cde2768e20b335b15deb8ebb717f3d4632bdd7da8caf0" name="opensips" arch="src">
  <version epoch="0" ver="3.2.5" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="bcf7dc72b40c45b8bbd347d29778bb1903f3792f94d6b90e6ea15e1d9b4ef54f" name="opensips" arch="src">
  <version epoch="0" ver="3.2.6" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="5cf92bb684f70c510791da362c7c5090c61e6edea9a17eeb153c099cc9536801" name="opensips" arch="src">
  <version epoch="0" ver="3.2.7" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="dbe34301562d20fc5122f81bce136f1e220d57a87ef23963996a96a6b074b364" name="opensips" arch="src">
  <version epoch="0" ver="3.2.8" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="d4a4b1b5d62824ef7823b2730225fe8f997829d0f9b833e6a47711ef5ed36824" name="opensips" arch="src">
  <version epoch="0" ver="3.2.9" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.10.0-1" date="1375185600">- Update to 1.10.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
</package>
<package pkgid="5de54d8177556a01e507dd2307efd897e083a1c7d0cef316eceb25e4625bbdee" name="opensips" arch="src">
  <version epoch="0" ver="3.3.0" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
</package>
<package pkgid="c8f4af601990e258575b7d231577ac9454c56fddcbe8e03e743e3d19a878307c" name="opensips" arch="src">
  <version epoch="0" ver="3.3.1" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
</package>
<package pkgid="fe0d9169aa818a2c1fd2d5599528bc634170213e9ed9a03747ff324c417417df" name="opensips" arch="src">
  <version epoch="0" ver="3.3.10" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.6-1" date="1687348800">- OpenSIPS minor stable release: 3.3.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.7-1" date="1693483200">- OpenSIPS minor stable release: 3.3.7-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.8-1" date="1697630400">- OpenSIPS minor stable release: 3.3.8-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.9-1" date="1703073600">- OpenSIPS minor stable release: 3.3.9-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.10-1" date="1708516800">- OpenSIPS minor stable release: 3.3.10-1</changelog>
</package>
<package pkgid="de3b6fdf950dfa42e183d0fe492b6e53a8c46d6df740e640c32d8376473b6a80" name="opensips" arch="src">
  <version epoch="0" ver="3.3.2" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
</package>
<package pkgid="3969edc02fd6f3007f679a1004c6ca0f755529cd5fd751fb724199c0c85d9fcf" name="opensips" arch="src">
  <version epoch="0" ver="3.3.3" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
</package>
<package pkgid="ecd75943a0980672235f95366b7ee5b54aa2dafcaec32abdf71eb20f250c5283" name="opensips" arch="src">
  <version epoch="0" ver="3.3.4" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
</package>
<package pkgid="0af4f261725ebf4da8093f85782e52f4970a7c866075c9914a43df1092bdd529" name="opensips" arch="src">
  <version epoch="0" ver="3.3.5" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
</package>
<package pkgid="ba11ad65f62780eb32b25d4e581ddd27d39680cac4dd89c1ada6ed4cef50cc69" name="opensips" arch="src">
  <version epoch="0" ver="3.3.6" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.6-1" date="1687348800">- OpenSIPS minor stable release: 3.3.6-1</changelog>
</package>
<package pkgid="8f8cb5a6e855b33e5b7fa48d0013e3cd1621c16e27d99478481f6e40ae00a51a" name="opensips" arch="src">
  <version epoch="0" ver="3.3.7" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.6-1" date="1687348800">- OpenSIPS minor stable release: 3.3.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.7-1" date="1693483200">- OpenSIPS minor stable release: 3.3.7-1</changelog>
</package>
<package pkgid="5b154dff2948b9ae9379224d466ce3f7bd353f4a7ec029bf4e25a6f91cb6f0ef" name="opensips" arch="src">
  <version epoch="0" ver="3.3.8" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.6-1" date="1687348800">- OpenSIPS minor stable release: 3.3.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.7-1" date="1693483200">- OpenSIPS minor stable release: 3.3.7-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.8-1" date="1697630400">- OpenSIPS minor stable release: 3.3.8-1</changelog>
</package>
<package pkgid="ae4b38a8f7482c04951c60789488efbad6efb4c9d41d1370fcc2cb74f79687de" name="opensips" arch="src">
  <version epoch="0" ver="3.3.9" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.6-1" date="1687348800">- OpenSIPS minor stable release: 3.3.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.7-1" date="1693483200">- OpenSIPS minor stable release: 3.3.7-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.8-1" date="1697630400">- OpenSIPS minor stable release: 3.3.8-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.3.9-1" date="1703073600">- OpenSIPS minor stable release: 3.3.9-1</changelog>
</package>
<package pkgid="8608f04de1dfa81f215f2b3fdbd9924ea78ecc95e7974c827f693df5a77ef1b1" name="opensips" arch="src">
  <version epoch="0" ver="3.4.0.beta" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 1.11.0-1" date="1395403200">- Update to 1.11.0</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.1.0-1" date="1426334400">- Specification updated for opensips 2.1
- Removed packages: auth_diameter, tlsops
- New packages: compression, emergency, fraud_detection,
  proto_sctp, proto_tls, proto_ws, rtpengine, topology_hiding</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
</package>
<package pkgid="cd34c23f96518bac05dfd1b280b88f480ae89f4489a2e2d68c21047c5155336d" name="opensips" arch="src">
  <version epoch="0" ver="3.4.0.rc1" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.2.0-1" date="1453291200">- Specification updated for opensips 2.2
- New packages: db_sqlite, clusterer, event_flatstore,
  event_virtual, proto_bin, proto_hep, proto_wss, sql_cacher
- Renamed packages: mysql -&gt; db_mysql, postgres -&gt; db_postgres,
  cpl-c -&gt; cpl_c</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.3.0-1" date="1488801600">- Specification updated for opensips 2.3
- New packages: event_routing, freeswitch, mid_registrar, sip_i, xml
- Enabled packages: cachedb_mongodb, lua
- Renamed packages: memcached -&gt; cachedb_memcached, redis -&gt; cachedb_redis,
  unixodbc -&gt; db_unixodbc, xmlrpc -&gt; mi_xmlrpc
- Added possibility to build unsupported modules (from obsolete .spec):
  cachedb_cassandra, cachedb_couchbase,
  cachedb_mongodb, osp, sngtc</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-rc1-1" date="1687262400">- OpenSIPS 3.4 new release candidate: 3.4.0-rc1-1</changelog>
</package>
<package pkgid="00c0e32abe9c52c65866375898531bcb6b8b8ac7a533b15d62d04d6e435b7604" name="opensips" arch="src">
  <version epoch="0" ver="3.4.1" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-rc1-1" date="1687262400">- OpenSIPS 3.4 new release candidate: 3.4.0-rc1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-1" date="1689768000">- OpenSIPS 3.4.0 stable release: 3.4.0-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.1-1" date="1693483200">- OpenSIPS minor stable release: 3.4.1-1</changelog>
</package>
<package pkgid="911f23eb92f5ac9977b04632c476b713b35bf9ce79dc42fd82a9709c933110c6" name="opensips" arch="src">
  <version epoch="0" ver="3.4.10" rel="1.el7_9"/>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.3-1" date="1703073600">- OpenSIPS minor stable release: 3.4.3-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.4-1" date="1708516800">- OpenSIPS minor stable release: 3.4.4-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.5-1" date="1713441600">- OpenSIPS minor stable release: 3.4.5-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.6-1" date="1718798400">- OpenSIPS minor stable release: 3.4.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.7-1" date="1721822400">- OpenSIPS minor stable release: 3.4.7-1</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.4.7-1" date="1724068800">- Replace deprecated dependency for radius modules</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.4.7-1" date="1724155200">- Add python-setuptools dependency</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.8-1" date="1724241600">- OpenSIPS minor stable release: 3.4.8-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.9-1" date="1729684800">- OpenSIPS minor stable release: 3.4.9-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.10-1" date="1734523200">- OpenSIPS minor stable release: 3.4.10-1</changelog>
</package>
<package pkgid="6db66c97abdf78aeb3241130f5393c5231b1f3427cebe019e5f952c34c642647" name="opensips" arch="src">
  <version epoch="0" ver="3.4.2" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-rc1-1" date="1687262400">- OpenSIPS 3.4 new release candidate: 3.4.0-rc1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-1" date="1689768000">- OpenSIPS 3.4.0 stable release: 3.4.0-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.1-1" date="1693483200">- OpenSIPS minor stable release: 3.4.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.2-1" date="1697630400">- OpenSIPS minor stable release: 3.4.2-1</changelog>
</package>
<package pkgid="09e6a2b6424d6871f0687bd74b57a29b808d48faaad4682ab9f6b3b95dd94cc1" name="opensips" arch="src">
  <version epoch="0" ver="3.4.3" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-rc1-1" date="1687262400">- OpenSIPS 3.4 new release candidate: 3.4.0-rc1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-1" date="1689768000">- OpenSIPS 3.4.0 stable release: 3.4.0-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.1-1" date="1693483200">- OpenSIPS minor stable release: 3.4.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.2-1" date="1697630400">- OpenSIPS minor stable release: 3.4.2-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.3-1" date="1703073600">- OpenSIPS minor stable release: 3.4.3-1</changelog>
</package>
<package pkgid="b70ffeb1a338fb4e639fc69af69e95a1a6b7391e9d447d3f2a7567f7136c9c0b" name="opensips" arch="src">
  <version epoch="0" ver="3.4.5" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-rc1-1" date="1687262400">- OpenSIPS 3.4 new release candidate: 3.4.0-rc1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-1" date="1689768000">- OpenSIPS 3.4.0 stable release: 3.4.0-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.1-1" date="1693483200">- OpenSIPS minor stable release: 3.4.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.2-1" date="1697630400">- OpenSIPS minor stable release: 3.4.2-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.3-1" date="1703073600">- OpenSIPS minor stable release: 3.4.3-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.4-1" date="1708516800">- OpenSIPS minor stable release: 3.4.4-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.5-1" date="1713441600">- OpenSIPS minor stable release: 3.4.5-1</changelog>
</package>
<package pkgid="a10cf2de98dcb8a32467d0fc84d358fc8a91d185c34f3f18e2120029b1d2460b" name="opensips" arch="src">
  <version epoch="0" ver="3.4.7" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-rc1-1" date="1687262400">- OpenSIPS 3.4 new release candidate: 3.4.0-rc1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.0-1" date="1689768000">- OpenSIPS 3.4.0 stable release: 3.4.0-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.1-1" date="1693483200">- OpenSIPS minor stable release: 3.4.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.2-1" date="1697630400">- OpenSIPS minor stable release: 3.4.2-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.3-1" date="1703073600">- OpenSIPS minor stable release: 3.4.3-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.4-1" date="1708516800">- OpenSIPS minor stable release: 3.4.4-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.5-1" date="1713441600">- OpenSIPS minor stable release: 3.4.5-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.6-1" date="1718798400">- OpenSIPS minor stable release: 3.4.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.7-1" date="1721822400">- OpenSIPS minor stable release: 3.4.7-1</changelog>
</package>
<package pkgid="5926fdc8a231e0c48b46f0996246c73fdc179332e03d1605f683ab264666e0b8" name="opensips" arch="src">
  <version epoch="0" ver="3.4.8" rel="1.el7_9"/>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.1-1" date="1693483200">- OpenSIPS minor stable release: 3.4.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.2-1" date="1697630400">- OpenSIPS minor stable release: 3.4.2-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.3-1" date="1703073600">- OpenSIPS minor stable release: 3.4.3-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.4-1" date="1708516800">- OpenSIPS minor stable release: 3.4.4-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.5-1" date="1713441600">- OpenSIPS minor stable release: 3.4.5-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.6-1" date="1718798400">- OpenSIPS minor stable release: 3.4.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.7-1" date="1721822400">- OpenSIPS minor stable release: 3.4.7-1</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.4.7-1" date="1724068800">- Replace deprecated dependency for radius modules</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.4.7-1" date="1724155200">- Add python-setuptools dependency</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.8-1" date="1724241600">- OpenSIPS minor stable release: 3.4.8-1</changelog>
</package>
<package pkgid="2d67019bb423d2bc154783341f8c9d75c5e63aa01fed494d1eb3a41721508397" name="opensips" arch="src">
  <version epoch="0" ver="3.4.9" rel="1.el7_9"/>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.2-1" date="1697630400">- OpenSIPS minor stable release: 3.4.2-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.3-1" date="1703073600">- OpenSIPS minor stable release: 3.4.3-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.4-1" date="1708516800">- OpenSIPS minor stable release: 3.4.4-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.5-1" date="1713441600">- OpenSIPS minor stable release: 3.4.5-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.6-1" date="1718798400">- OpenSIPS minor stable release: 3.4.6-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.7-1" date="1721822400">- OpenSIPS minor stable release: 3.4.7-1</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.4.7-1" date="1724068800">- Replace deprecated dependency for radius modules</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.4.7-1" date="1724155200">- Add python-setuptools dependency</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.8-1" date="1724241600">- OpenSIPS minor stable release: 3.4.8-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.4.9-1" date="1729684800">- OpenSIPS minor stable release: 3.4.9-1</changelog>
</package>
<package pkgid="9f2c6106cbea907a154eb53c518dd6241ead33a3a8512282ec09c86858022d30" name="opensips" arch="src">
  <version epoch="0" ver="3.5.0" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 2.4.0-1" date="1522238400">- Specification updated for opensips 2.4
- New packages: event_jsonrpc, jsonrpc, siprec</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.6.0-1" date="1715256000">- Specification updated for opensips 3.6</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.5.0-1" date="1716033600">- Specification updated for opensips 3.5
- New modules: aka_av_diameter, auth_aka, mqueue, presence_reginfo, proto_ipsec, pua_reginfo</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.0-1" date="1721908800">- OpenSIPS minor stable release: 3.5.0-1</changelog>
</package>
<package pkgid="629cdc0c4f15b10c4c1c27afa3d94467bf05019a4403f89665480ea7211b4c7d" name="opensips" arch="src">
  <version epoch="0" ver="3.5.1" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.0.0-1" date="1554984000">- Specification updated for opensips 3.0
- Package names and layout were changed similar to debian packaging</changelog>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.5.0-1" date="1716033600">- Specification updated for opensips 3.5
- New modules: aka_av_diameter, auth_aka, mqueue, presence_reginfo, proto_ipsec, pua_reginfo</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.0-1" date="1721908800">- OpenSIPS minor stable release: 3.5.0-1</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.5.0-1" date="1724068800">- Replace deprecated dependency for radius modules</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.1-1" date="1724241600">- OpenSIPS minor stable release: 3.5.1-1</changelog>
</package>
<package pkgid="9473201bc2c5443c3be271bde4d2c13d2e965ad0c39cacf72656b30492c1cf2d" name="opensips" arch="src">
  <version epoch="0" ver="3.5.2" rel="1.el7_9"/>
  <changelog author="Nick Altmann &lt;nick.altmann@gmail.com&gt; - 3.1.0-1" date="1581422400">- Specification updated for opensips 3.1
- New modules: b2b_logic_xml, callops, media_exchange, presence_dfks,
  qrouting, rabbitmq_consumer, rate_cacher, stir_shaken, uuid
- New package: stir-shaken-module
- Obsoleted modules: seas, sms
- Renamed: event_jsonrpc -&gt; event_stream
- Removed: opensipsunix, osipsconsole, opensipsctl, opensipsdbctl</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.5.0-1" date="1716033600">- Specification updated for opensips 3.5
- New modules: aka_av_diameter, auth_aka, mqueue, presence_reginfo, proto_ipsec, pua_reginfo</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.0-1" date="1721908800">- OpenSIPS minor stable release: 3.5.0-1</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.5.0-1" date="1724068800">- Replace deprecated dependency for radius modules</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.1-1" date="1724241600">- OpenSIPS minor stable release: 3.5.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.2-1" date="1729684800">- OpenSIPS minor stable release: 3.5.2-1</changelog>
</package>
<package pkgid="113d5f0d62451a2c43d58dc0c2dbaa43436f542291b6db52b4506f0925153b24" name="opensips" arch="src">
  <version epoch="0" ver="3.5.3" rel="1.el7_9"/>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.2.0-1" date="1614340800">- New modules: prometheus</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.2.0-1" date="1622116800">- Specification updated for opensips 3.2
- New modules: aaa_diameter, b2b_logic, event_kafka, prometeus, rtp_relay, tls_openssl, tls_wolfssl
- New packages: aaa-diameter-module, kafka-module, prometeus-module, tls-openssl-module, tls-wolfssl-module
- Obsoleted modules: b2b_logic_xml</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.3.0-1" date="1652875200">- Specification updated for opensips 3.3
- New modules: b2b_sdp_demux, msrp_gateway, msrp_relay, msrp_ua, proto_msrp, status_report, tcp_mgm
- New packages: msrp-modules
- Removed modules: b2b_logic_xml
- Removed packages: b2bua-module</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.4.0-1" date="1684411200">- Specification updated for opensips 3.4</changelog>
  <changelog author="Nick Altmann &lt;nick@altmann.pro&gt; - 3.5.0-1" date="1716033600">- Specification updated for opensips 3.5
- New modules: aka_av_diameter, auth_aka, mqueue, presence_reginfo, proto_ipsec, pua_reginfo</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.0-1" date="1721908800">- OpenSIPS minor stable release: 3.5.0-1</changelog>
  <changelog author="Razvan Crainea &lt;razvan@opensips.org&gt; - 3.5.0-1" date="1724068800">- Replace deprecated dependency for radius modules</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.1-1" date="1724241600">- OpenSIPS minor stable release: 3.5.1-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.2-1" date="1729684800">- OpenSIPS minor stable release: 3.5.2-1</changelog>
  <changelog author="Liviu Chircu &lt;liviu@opensips.org&gt; - 3.5.3-1" date="1734523200">- OpenSIPS minor stable release: 3.5.3-1</changelog>
</package>
<package pkgid="6983ae78961c1452718af5c16c90b2e9f67da0b3c4d52a71d2635e461ad6c1e9" name="pcapsipdump" arch="src">
  <version epoch="0" ver="0.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.2-1" date="1479988800">- First version of this spec file</changelog>
</package>
<package pkgid="59e3f0cc8804eec83f1457373da569b7c375706e53ff435802267ed5da7a9d01" name="pcapsipdump" arch="src">
  <version epoch="0" ver="0.2" rel="2.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.2-1" date="1479988800">- First version of this spec file</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voidpomain.io&gt; - 0.2-2" date="1687953600">- Added patch with many improvements</changelog>
</package>
<package pkgid="9b86fd9103dccd7b5b3ec66a20700ea0cbba83524935e52d2ead2faae8470a80" name="perl-Geo-METAR" arch="src">
  <version epoch="0" ver="1.15" rel="27.el7_9"/>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 1.15-18" date="1409140800">- Perl 5.20 rebuild</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 1.15-19" date="1433332800">- Perl 5.22 rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.15-20" date="1434628800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 1.15-21" date="1444824000">- Specify all dependencies</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.15-22" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.15-23" date="1458302400">- Modernize spec file</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 1.15-24" date="1463313600">- Perl 5.24 rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.15-25" date="1486814400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Jitka Plesnikova &lt;jplesnik@redhat.com&gt; - 1.15-26" date="1496577600">- Perl 5.26 rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.15-27" date="1501156800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
</package>
<package pkgid="f3820dba914a72922e7a0c5b9fffb9ead5e109cbeadc207546c309a209515ae1" name="perl-Jcode" arch="src">
  <version epoch="0" ver="2.07" rel="15.el7_9"/>
  <changelog author="Marcela Ma&#x161;l&#xE1;&#x148;ov&#xE1; &lt;mmaslano@redhat.com&gt; 2.07-6" date="1308139200">- Perl mass rebuild</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; 2.07-7" date="1326369600">- Add buildreqs for perl core modules that might be dual-lived
- Nobody else likes macros for commands
- Use patch rather than scripted edit to fix encoding of Nihongo.pod</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; 2.07-8" date="1335441600">- Don't need to remove empty directories from buildroot
- Don't need to run test suite with LC_ALL=C
- BR: perl(Data::Dumper)
- Use search.cpan.org source URL</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 2.07-9" date="1339416000">- Perl 5.16 rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.07-10" date="1342785600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.07-11" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 2.07-12" date="1374062400">- Perl 5.18 rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.07-13" date="1375531200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.07-14" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Paul Howarth &lt;paul@city-fan.org&gt; - 2.07-15" date="1407758400">- Specify all dependencies
- Drop %defattr, redundant since rpm 4.4</changelog>
</package>
<package pkgid="ef41a835b09d9f5ea0d274d10c97dba8dcf2fea0aa67ee8fc47755cb1371bca2" name="perl-generators" arch="src">
  <version epoch="0" ver="1.08" rel="6.el7_9"/>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.08-4" date="1466596800">- Provide this dummy package to ease porting Fedora packages to EPEL</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.08-5" date="1503921600">- Provide perl-interpreter (bug #1410347)</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.08-6" date="1503921601">- Add epoch to perl-interpreter version.</changelog>
</package>
<package pkgid="39d17b40f45097bbaa25f687a933b6e2bc40e7d7bbd0422566d83e089923cb54" name="perl-generators" arch="src">
  <version epoch="0" ver="1.08" rel="7.el7_9"/>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.08-4" date="1466596800">- Provide this dummy package to ease porting Fedora packages to EPEL</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.08-5" date="1503921600">- Provide perl-interpreter (bug #1410347)</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.08-6" date="1503921601">- Add epoch to perl-interpreter version.</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 1.08-7" date="1546862400">- Remove perl-interpreter package (bug #1663304)</changelog>
</package>
<package pkgid="003f44fef31219f1c91283b760a347e7a37df17d61d3e41cb402287f872fc1e8" name="php-curve25519" arch="src">
  <version epoch="0" ver="0.1.git20160624" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.1.git20160624-1" date="1468843200">- First version of this spec</changelog>
</package>
<package pkgid="6d13816f6e2fff6eade3ded5a70de3ef915a2d3217b0455ee2938680e052490f" name="php-extras" arch="src">
  <version epoch="0" ver="5.4.16" rel="4.el7_9"/>
  <changelog author="Jon Ciesla &lt;limb@jcomserv.net&gt; - 5.3.2-1" date="1292760000">- Update for EL-6.</changelog>
  <changelog author="Nick Bebout &lt;nb@fedoraproject.org&gt; - 5.3.2-2" date="1292932800">- Disable php-readline for now so we can get the rest built</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 5.3.2-3" date="1293019200">- drop readline (now in the main php package)
- drop mhash (no more provided in php, hash used instead)
- spec file cleanup</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 5.3.3-1" date="1315915200">- update to 5.3.3
- avoid explicit php dependencies in favour of php-api (#737956)</changelog>
  <changelog author="Remi Collet &lt;rcollet@redhat.com&gt; - 5.4.16-0" date="1389700800">- update to 5.4.16
- add virtual provides for all provided extensions
- with imap, mssql, tidy</changelog>
  <changelog author="Remi Collet &lt;rcollet@redhat.com&gt; - 5.4.16-0.1" date="1389700801">- add mcrypt</changelog>
  <changelog author="Remi Collet &lt;rcollet@redhat.com&gt; - 5.4.16-1" date="1389873600">- add interbase
- add workaround for https://bugzilla.redhat.com/1054659
- minimal load test for all extensions</changelog>
  <changelog author="Remi Collet &lt;rcollet@redhat.com&gt; - 5.4.16-2" date="1390392000">- drop interbase patch</changelog>
  <changelog author="Remi Collet &lt;rcollet@redhat.com&gt; - 5.4.16-3" date="1391083200">- mcrypt upstream security fix</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 5.4.16-4" date="1457524800">- add dblib patch (#1266177)</changelog>
</package>
<package pkgid="c5eb26686942d5cd2584081a0f368bd82088cae16f594d76904514234415cec4" name="php-pecl-gearman" arch="src">
  <version epoch="0" ver="1.1.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.2-1" date="1572004800">- First version of this script</changelog>
</package>
<package pkgid="c34e9d7b17546a1b9f232ea37c3056596a87146f198bafaf2ed69dd26461baf0" name="php-pecl-inotify" arch="src">
  <version epoch="0" ver="0.1.6" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.1.6-1" date="1366632000">- first version</changelog>
</package>
<package pkgid="9bd7ff94f2700f37c7e3d7574fe18a554e478243bb98c86fbe62b8fb0ca11ce6" name="php-pecl-libevent" arch="src">
  <version epoch="0" ver="0.1.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; 0.0.5-1" date="1367236800">- first version of this package</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; 0.1.0-1" date="1381147200">- updated to version 0.1.0</changelog>
</package>
<package pkgid="4e3fbdbc290d48803eb0c21116b78cbe36366afe8534550ef4b9f78208228547" name="php-pecl-ncurses" arch="src">
  <version epoch="0" ver="1.0.2" rel="1.el7_9"/>
  <changelog author="Remi Collet &lt;Fedora@FamilleCollet.com&gt; 1.0.0-1" date="1247400000">- initial RPM (for php 5.3.0)
- ncurses-1.0.0-php53.patch</changelog>
  <changelog author="Remi Collet &lt;Fedora@FamilleCollet.com&gt; 1.0.0-2" date="1247400001">- add %check for minimal test.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0.0-3" date="1248609600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;Fedora@FamilleCollet.com&gt; 1.0.1-1" date="1261224000">- update to 1.0.1
- enable wide char support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; 1.0.2-1" date="1381147200">- update to 1.0.2</changelog>
</package>
<package pkgid="80971083fcfc43e609c397d694fd371f26f5157b85962fa24e5aa31449b4bab1" name="php-pecl-proctitle" arch="src">
  <version epoch="0" ver="0.1.2" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; 0.1.2-1" date="1366632000">- first version</changelog>
</package>
<package pkgid="6c9a2f44722b621ccbf16643cd13cc1902dfa3238224f0a94bda85eef1f531b1" name="php-pecl-rrd" arch="src">
  <version epoch="0" ver="1.1.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.3-1" date="1477051200">- First version of this spec</changelog>
</package>
<package pkgid="8c3ccf01a1add36d8dfc6ad53fe9f02d9cb832373640db26a66d8108593e5bb3" name="php-pecl-runkit" arch="src">
  <version epoch="0" ver="1.0.4" rel="0.3.GIT8c73eaf.el7_9"/>
  <changelog author="Pavel Alexeev &lt;Pahan@Hubbitus.info&gt; - 0.9-10.CVS20090215" date="1237291200">- Remi Collet notes in Fedora review:
- Rename back %{peclName}.xml to %{name}.xml :)
- Set %defattr(-,root,root,-) (was %defattr(644,root,root,755))
- Make the %post/%postun scriptlets silent</changelog>
  <changelog author="Pavel Alexeev &lt;Pahan@Hubbitus.info&gt; - 0.9-11.CVS20090215" date="1238932800">- By suggestion in the bug https://fedorahosted.org/fedora-infrastructure/ticket/1298 try remove version specifications in requires:
	Turn BuildRequires: php-pear &gt;= 1.4.7, php-devel &gt;= 5.0.0 to just BuildRequires: php-pear, php-devel
- Add more magick in Release tag: 11%{?CVS:.CVS%{CVS}}%{?dist}</changelog>
  <changelog author="Pavel Alexeev &lt;Pahan@Hubbitus.info&gt; - 0.9-12.CVS20090215" date="1248955200">- Apply patches only on PHP&gt;=5.3.0. (Bug: https://bugzilla.redhat.com/show_bug.cgi?id=513096 )</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.9-13.CVS20090215" date="1297252800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Pavel Alexeev &lt;Pahan@Hubbitus.info&gt; - 0.9-14.CVS20090215" date="1312286400">- Fix FBFS on Fedora 16(17 rawhide) and rpm 4.9 - bz#715709</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.9-15.CVS20090215" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.9-16.CVS20090215" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Pavel Alexeev &lt;Pahan@Hubbitus.info&gt; - 1.0.4-0.1.GIT8c73eaf" date="1346673600">- New upstream url - https://bugs.php.net/bug.php?id=61189, continue developing, new version, switch to git SCM.
- Fix compilation error: https://github.com/zenovich/runkit/issues/26#issuecomment-8268795</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0.4-0.2.GIT8c73eaf" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;rcollet@redhat.com&gt; - 1.0.4-0.3.GIT8c73eaf" date="1363953600">- rebuild for http://fedoraproject.org/wiki/Features/Php55</changelog>
</package>
<package pkgid="e47511d7c38c6a7f6934c15a66ee555f86fac2cf34cb1732c8396c36fc6307c6" name="php-pecl-xcache" arch="src">
  <version epoch="0" ver="3.2.0" rel="1.el7_9"/>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 3.0.0-1" date="1351684800">- new major version
- drop xcache-coverager subpackage
- xcache-admin now provides cacher, coverager and diagnosis
- run unit tests provided by upstream</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 3.0.0-1.1" date="1353585600">- upstream have fixed the sources (review #859504)</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 3.0.1-1" date="1358424000">- bugfixes version</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.2-1" date="1381147200">- updated to version 3.0.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.3-1" date="1381147201">- updated to version 3.0.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.0.4-1" date="1383566400">- updated to version 3.0.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.0-1" date="1383566401">- updated to version 3.1.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.1-1" date="1450699200">- updated to version 3.1.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.1.2-1" date="1450699201">- updated to version 3.1.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.2.0-1" date="1450699202">- updated to version 3.2.0</changelog>
</package>
<package pkgid="b5da106db0ea6d62153b1a5539f723a2f54c38fa46f8f896f5ec1142e12725b8" name="php-pecl-xdebug" arch="src">
  <version epoch="0" ver="2.4.1" rel="1.el7_9"/>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 2.2.2-1" date="1364126400">- update to 2.2.2 (stable)
- run buildconf for aarch64 support #926329
- modernize spec</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 2.2.3-1" date="1369224000">- Update to 2.2.3</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 2.2.3-2" date="1373976000">- adapt for SCL</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.2.3-3" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 2.2.4-1" date="1393761600">- Update to 2.2.4 (stable)
- move documentation in pecl_docdir
- cleanups</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 2.2.4-2" date="1398254400">- add numerical prefix to extension configuration file
- drop uneeded full extension path</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 2.2.5-1" date="1398859200">- Update to 2.2.5 (stable)</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 2.2.6-1" date="1416139200">- Update to 2.2.6 (stable)</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 2.2.7-1" date="1421928000">- Update to 2.2.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 2.4.1-1" date="1527076800">- Update to 2.4.1</changelog>
</package>
<package pkgid="cdb8d3032978134cb28102918c3234e6e46b43b604783a9cf91527291ff3daa6" name="php-pecl-zmq" arch="src">
  <version epoch="0" ver="1.1.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.3-1" date="1572264000">- First version of this script</changelog>
</package>
<package pkgid="7d9c7803ec582373d92baeaa6de0051eb18d3e78b06e623dea38d2438e32cea7" name="php-protobuf" arch="src">
  <version epoch="0" ver="0.11.0" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.11.0-1" date="1468843200">- First version of this spec</changelog>
</package>
<package pkgid="663df5f557d5942a5d270d128fc9c51dea105356d3c8845be0055150e82fb378" name="php-runkit" arch="src">
  <version epoch="0" ver="1.0.4" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; 0.0.5-1" date="1367236800">- first version of this package</changelog>
</package>
<package pkgid="807cf27c23fa465d6c9d38296718d7b0282c2573b25b1dcc5862604b47bf6c61" name="php-suhosin" arch="src">
  <version epoch="2" ver="0.9.38" rel="1.el7_9"/>
  <changelog author="Denis Frolov &lt;d.frolov81@mail.ru&gt; - 0.9.29-3" date="1268049600">- Rebuild for php 5.3.2</changelog>
  <changelog author="Denis Frolov &lt;d.frolov81@mail.ru&gt; - 0.9.30-1" date="1269777600">- Update to 0.9.30</changelog>
  <changelog author="Denis Frolov &lt;d.frolov81@mail.ru&gt; - 0.9.31-1" date="1269864000">- Update to 0.9.31</changelog>
  <changelog author="Denis Frolov &lt;d.frolov81@mail.ru&gt; - 0.9.32.1-1" date="1283428800">- Update to 0.9.32.1</changelog>
  <changelog author="Denis Frolov &lt;d.frolov81@mail.ru&gt; - 0.9.33.1-1" date="1363780800">- Update to 0.9.33.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.9.35-1" date="1399291200">- Update to 0.9.35</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.9.36-1" date="1406635200">- Update to 0.9.36</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.9.37-1" date="1450699200">- Update to 0.9.37</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.9.37.1-1" date="1450699201">- Update to 0.9.37.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.9.38-1" date="1450699202">- Update to 0.9.38</changelog>
</package>
<package pkgid="7dea26678528096bbc1e3308f7288003d149d94ca38a07ce885b1d636cd6ccff" name="php-zend-loader" arch="src">
  <version epoch="0" ver="3.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.3-1" date="1451822400">- First build script</changelog>
</package>
<package pkgid="7912a0756c7b0bc04bd2d86f496375928fe4cb7753a1316da3b789f73a35983a" name="php-zend-optimizer" arch="src">
  <version epoch="0" ver="3.3.9" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.3.9-1" date="1282824000">- First build script</changelog>
</package>
<package pkgid="6e75de0f628bc2619e68f55ab95cbe9b0924844dab63432c9522d688e80edfdd" name="pjproject" arch="src">
  <version epoch="0" ver="2.3" rel="6.el7_9"/>
  <changelog author="Jared Smith &lt;jsmiht@fedoraproject.org&gt; - 2.2-1" date="1393588800">- Update to upstream 2.2 release</changelog>
  <changelog author="Jared Smith &lt;jsmiht@fedoraproject.org&gt; - 2.2-2" date="1393588801">- Rebase a patch to simple eliminate the third_party directory
- Add a patch to fix up the .pc file</changelog>
  <changelog author="Jared Smith &lt;jsmiht@fedoraproject.org&gt; - 2.2-3" date="1393588802">- Fix the location of the .so files
- Add a massive patch to fix the incorrect FSF address</changelog>
  <changelog author="Jared Smith &lt;jsmiht@fedoraproject.org&gt; - 2.2-4" date="1394366400">- Instead of deleting the empty file pj/config_site.h, make it non-empty</changelog>
  <changelog author="Jared Smith &lt;jsmiht@fedoraproject.org&gt; - 2.2.1-1" date="1403265600">- Update to upstream 2.2.1 release</changelog>
  <changelog author="Jared Smith &lt;jsmiht@fedoraproject.org&gt; - 2.3-1" date="1410350400">- Update to upstream 2.3 release</changelog>
  <changelog author="Jared Smith &lt;jsmiht@fedoraproject.org&gt; - 2.3-2" date="1410350401">- Disable video support, and specifically tell it not to use libyuv,
  as the version of libyuv in Fedora is too old
- Disable ilbc codec support, as it is not needed</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 2.3-3" date="1413374400">- Add IPv6 support</changelog>
  <changelog author="Jared Smith &lt;jsmith@fedoraproject.org&gt; - 2.3-5" date="1414324800">- Fix endianness support on ARM platform</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 2.3-6" date="1415966400">- rebuild against new libsrtp</changelog>
</package>
<package pkgid="53dc0eaca6f9e1910f3d66e51af686c4c44d07050b53516111942d6b842c912b" name="portaudio" arch="src">
  <version epoch="0" ver="19" rel="9.el7_9"/>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 18.1-8" date="1156766400">- FC6 rebuild.</changelog>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 19-1" date="1182254400">- Update to "stable" v19_061121.
- Switch virtual devel provide to a real sub-package.
- Update spec to match build changes from custom Makefile to autotools.
- Include new pkgconfig file and require pkgconfig from the devel package.
- Add ldconfig calls now that we have a versionned shared library.
- Rebuild against alsa-lib and jack-audio-connection-kit.
- Build doxygen documentation and include it in the devel package.</changelog>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 19-2" date="1186315200">- Update License field.</changelog>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 19-3" date="1187784000">- Rebuild for new BuildID feature.</changelog>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 19-4" date="1197288000">- Include portaudiocpp library and headers (#413681).</changelog>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 19-5" date="1202040000">- Update to "stable" v19_20071207.
- Rebuild against latest jack in rawhide (#430672).
- Backport update to F8 too (#431266).</changelog>
  <changelog author="Hans de Goede &lt;hdegoede@redhat.com&gt; 19-6" date="1228651200">- Add a patch by Kevin Kofler to make non mmap alsa (and thus pulseaudio) work
  (bz 445644)</changelog>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 19-7" date="1229947200">- Add Doxyfile patch to remove date in footer and fix multilib (#342931).</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 19-8" date="1235649600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 19-9" date="1248609600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
</package>
<package pkgid="6229cfc6c4f2786e46583573503151023bb032b62a413ac8b5829611431cb1a0" name="proxychains-ng" arch="src">
  <version epoch="0" ver="4.11" rel="1.el7_9"/>
  <changelog author="Pranav Kant &lt;pranav913@gmail.com&gt; 4.8.1-4" date="1426507200">- Install .so file with executable flags
- Replace old optflags patch with corrected patch</changelog>
  <changelog author="Pranav Kant &lt;pranav913@gmail.com&gt; 4.8.1-5" date="1426507201">- Remove .so versioning</changelog>
  <changelog author="Pranav Kant &lt;pranav913@gmail.com&gt; 4.8.1-6" date="1426593600">- Remove legacy script - proxyresolv4
- Move .so file to application-specific directory</changelog>
  <changelog author="Pranav Kant &lt;pranav913@gmail.com&gt; 4.8.1-7" date="1431086400">- Fixed fsf patch from upstream
- Added Obsoletes</changelog>
  <changelog author="Pranav Kant &lt;pranvk@fedoraproject.org&gt; 4.8.1-8" date="1432209600">- Fix CVE-2015-3887</changelog>
  <changelog author="Pranav Kant &lt;pranvk@fedoraproject.org&gt; 4.8.1-9" date="1434283200">- Fix source URL</changelog>
  <changelog author="Pranav Kant &lt;pranvk@fedoraproject.org&gt; 4.10-1" date="1434283201">- Update to 4.10</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 4.10-2" date="1434628800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.10-3" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Pranav Kant &lt;pranvk@fedoraproject.org&gt; 4.11-1" date="1455364800">- Update to 4.11</changelog>
</package>
<package pkgid="9cc6c66aa0cd151dc217edc726520bdf72e8c8baf94d0159edbb9f7c8012fa23" name="puppet" arch="src">
  <version epoch="0" ver="5.5.10" rel="4.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 5.5.1-2" date="1531483200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Ha&#xEF;kel Gu&#xE9;mar &lt;hguemar@fedoraproject.org&gt; - 5.5.6-1" date="1540987200">- Upstream 5.5.6
- Fix issues with RHEL &gt; 7</changelog>
  <changelog author="Bogdan Dobrelya &lt;bdobreli@redhat.com&gt; - 5.5.6-3" date="1548158400">- Use systemd_ordering macro</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 5.5.6-4" date="1549108800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Bogdan Dobrelya &lt;bdobreli@redhat.com&gt; - 5.5.6-5" date="1550404800">- Revert use of systemd_ordering macro</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 5.5.6-6" date="1551960000">- Split off headless subpackage, based on idea from Bogdan Dobrelya</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 5.5.10-1" date="1551960001">- 5.5.10</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 5.5.10-2" date="1551960002">- Move reqs to headless</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 5.5.10-3" date="1551960003">- Move sysconfdirs to headless too</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 5.5.10-4" date="1552478400">- Minor clean up</changelog>
</package>
<package pkgid="51311f25d5cf75f8d4777e45bf2ec0cb4ec76c7938510620d5d8c52306372ef8" name="puppet-firewalld" arch="src">
  <version epoch="0" ver="0.1.3" rel="1.el7_9"/>
  <changelog author="Jiri Popelka &lt;jpopelka@redhat.com&gt; - 0.1-1" date="1405080000">- initial version</changelog>
  <changelog author="Jiri Popelka &lt;jpopelka@redhat.com&gt; - 0.1.1-1" date="1405339200">- improved metadata</changelog>
  <changelog author="Jiri Popelka &lt;jpopelka@redhat.com&gt; - 0.1.1-2" date="1405512000">- fixed source URL</changelog>
  <changelog author="Jiri Popelka &lt;jpopelka@redhat.com&gt; - 0.1.2-1" date="1407153600">- 0.1.2</changelog>
  <changelog author="Jiri Popelka &lt;jpopelka@redhat.com&gt; - 0.1.3-1" date="1408536000">- 0.1.3</changelog>
</package>
<package pkgid="50a5c60a13bb9cf5f24528aab8517b190a1dd8a22bab607c4a85e2f85d44d72a" name="puppetlabs-stdlib" arch="src">
  <version epoch="0" ver="4.25.1" rel="1.el7_9"/>
  <changelog author="Andrea Veri &lt;averi@fedoraproject.org&gt; - 4.2.1-1.20140510git08b00d9" date="1399723200">- New upstream release.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 4.2.1-2.20140510git08b00d9" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Andrea Veri &lt;averi@fedoraproject.org&gt; - 4.5.1-1.20150121git7a91f20" date="1421841600">- New upstream release. (Fixes CVE-2015-1029, Red Hat's BZ #1182578)</changelog>
  <changelog author="Andrea Veri &lt;averi@fedoraproject.org&gt; - 4.5.1-2.20150121git7a91f20" date="1423051200">- Make sure metadata.json gets installed correctly for Puppet to actually
  recognize the module release version. Thanks Simon Lukasik for the patch.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 4.5.1-3.20150121git7a91f20" date="1434628800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.5.1-4.20150121git7a91f20" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.5.1-5.20150121git7a91f20" date="1486814400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.5.1-6.20150121git7a91f20" date="1501156800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.5.1-7.20150121git7a91f20" date="1518177600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Andrea Veri &lt;averi@fedoraproject.org&gt; - 4.25.1-1" date="1525953600">- New upstream release.</changelog>
</package>
<package pkgid="d9d08f0edd6b6805284c3244887dd0291c88bfe773d0819266f6dd20ca21a982" name="python-BeautifulSoup" arch="src">
  <version epoch="1" ver="3.2.1" rel="7.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:3.2.0-2" date="1297166400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:3.2.0-3" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 1:3.2.0-4" date="1328270400">- Add Python 3 package based on BeautifulSoup 4</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 1:3.2.1-1" date="1330344000">- 3.2.1
- Update bs4 to 4.0.0b8
- Add dep on lxml and html5lib for bs4</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:3.2.1-2" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 1:3.2.1-3" date="1343736000">- Beautiful Soup 4 has been moved to package python-beautifulsoup4</changelog>
  <changelog author="David Malcolm &lt;dmalcolm@redhat.com&gt; - 1:3.2.1-4" date="1344081600">- rebuild for https://fedoraproject.org/wiki/Features/Python_3.3</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:3.2.1-5" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:3.2.1-6" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 1:3.2.1-7" date="1385553600">- Add provide for python-beautifulsoup</changelog>
</package>
<package pkgid="2b20ee209c26c672089810127da6c218acb54cf83496d72890e15bee480e9065" name="python-LEPL" arch="src">
  <version epoch="0" ver="5.1.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 5.1.3-1" date="1654171200">- First spec version</changelog>
</package>
<package pkgid="00bdf14108acdc3518f4347e4fe059f7b77056fba88b522e4ac5f3a5b8378a94" name="python-colorama" arch="src">
  <version epoch="0" ver="0.3.2" rel="3.el7_9"/>
  <changelog author="Matthias Runge &lt;mrunge@redhat.com&gt; - 0.2.5-1" date="1363953600">- update to 0.2.5 (rhbz#913431)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.2.5-2" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Matthias Rugne &lt;mrunge@redhat.com&gt; - 0.2.7-1" date="1380542400">- uddate to 0.2.7 (rhbz#1010924)</changelog>
  <changelog author="Matthias Runge &lt;mrunge@redhat.com&gt; - 0.2.7-2" date="1394625600">- introduce python3 package (rhbz#1075410)</changelog>
  <changelog author="Kalev Lember &lt;kalevlember@gmail.com&gt; - 0.2.7-3" date="1401278400">- Rebuilt for https://fedoraproject.org/wiki/Changes/Python_3.4</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.2.7-4" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Lubomir Rintel &lt;lkundrak@v3.sk&gt; - 0.2.7-5" date="1406289600">- Skip the python3 %files section if we don't build the python3 package</changelog>
  <changelog author="Matthias Runge &lt;mrunge@redhat.com&gt; - 0.3.2-1" date="1409918400">- update to 0.3.2 (rhbz#1090014)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.3.2-2" date="1434628800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.3.2-3" date="1447156800">- Rebuilt for https://fedoraproject.org/wiki/Changes/python3.5</changelog>
</package>
<package pkgid="4cabd4e09a78e6b1176e815d14e6dc291f24587e1c79ff9f1ca7ebb3376188f7" name="python-configargparse" arch="src">
  <version epoch="0" ver="0.11.0" rel="1.el7_9"/>
  <changelog author="Fabian Affolter &lt;mail@fabian-affolter.ch&gt; - 0.9.3-1" date="1423137600">- Initial package</changelog>
  <changelog author="Fabian Affolter &lt;mail@fabian-affolter.ch&gt; - 0.9.3-2" date="1446206400">- Update macros</changelog>
  <changelog author="Fabian Affolter &lt;mail@fabian-affolter.ch&gt; - 0.9.3-3" date="1446724800">- Remove old parts</changelog>
  <changelog author="Kalev Lember &lt;klember@redhat.com&gt; - 0.9.3-4" date="1447329600">- Rebuilt for https://fedoraproject.org/wiki/Changes/python3.5</changelog>
  <changelog author="Robert Buchholz &lt;rbu@goodpoint.de&gt; - 0.9.3-5" date="1449144000">- epel7: Only build python2 package</changelog>
  <changelog author="Nick Bebout &lt;nb@fedoraproject.org&gt; - 0.10.0-1" date="1453377600">- Update to 0.10.0 for Let's Encrypt dep</changelog>
  <changelog author="Fabian Affolter &lt;mail@fabian-affolter.ch&gt; - 0.11.0-1" date="1479211200">- Update to new upstream version 0.11.0 (rhbz#1382975)</changelog>
</package>
<package pkgid="e588612e2afea642367d4b0725cee5437dff091f8d43b6d22598fe965eed7da9" name="python-distro" arch="src">
  <version epoch="0" ver="1.2.0" rel="3.el7_9"/>
  <changelog author="Miro Hron&#x10D;ok &lt;mhroncok@redhat.com&gt; - 1.0.0-7" date="1482148800">- Rebuild for Python 3.6</changelog>
  <changelog author="Igor Gnatenko &lt;i.gnatenko.brain@gmail.com&gt; - 1.0.1-1" date="1483272000">- Update to 1.0.1
- Provide only one copy of $bindir/distro
- Cleanups in spec</changelog>
  <changelog author="Miroslav Such&#xFD; &lt;msuchy@redhat.com&gt; 1.0.1-2" date="1483444800">- soft deps on lsb_release</changelog>
  <changelog author="Miroslav Such&#xFD; &lt;msuchy@redhat.com&gt; 1.0.2-1" date="1485259200">- update to 1.0.2
- 1415667 - require python-argparse on EL6</changelog>
  <changelog author="Miroslav Such&#xFD; &lt;msuchy@redhat.com&gt; 1.0.2-2" date="1485259201">- add license macro for el6</changelog>
  <changelog author="Miroslav Such&#xFD; &lt;msuchy@redhat.com&gt; 1.0.2-3" date="1485259202">- typo in license macro</changelog>
  <changelog author="Miroslav Such&#xFD; &lt;msuchy@redhat.com&gt; 1.0.3-1" date="1490011200">- rebase to 1.0.3</changelog>
  <changelog author="Miroslav Such&#xFD; &lt;msuchy@redhat.com&gt; 1.2.0-1" date="1514894400">- run tests
- rebase to distro 1.2.0</changelog>
  <changelog author="Scott K Logan &lt;logans@cottsay.net&gt; 1.2.0-2" date="1549281600">- add python 3 support for EL7</changelog>
  <changelog author="Miro Hron&#x10D;ok &lt;mhroncok@redhat.com&gt; - 1.2.0-3" date="1556107200">- Add explicit conflicts for clean update path of /usr/bin/distro</changelog>
</package>
<package pkgid="824732e0ffe121156e40ba64d8ca0f05934cd54e33630e884773ebdb600a8dcf" name="python-empy" arch="src">
  <version epoch="0" ver="3.3.2" rel="2.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.3-10" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.3-11" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Orion Poplawski &lt;orion@cora.nwra.com&gt; - 3.3.1-1" date="1390564800">- Update to 3.3.1</changelog>
  <changelog author="Filipe Rosset &lt;rosset.filipe@gmail.com&gt; - 3.3.2-1" date="1394193600">- Update to 3.3.2</changelog>
  <changelog author="Filipe Rosset &lt;rosset.filipe@gmail.com&gt; - 3.3.2-2" date="1394280000">- Fix packaging issues, spec cleanup
- Now compatible with python2 and python3 rhbz #1073768</changelog>
</package>
<package pkgid="17f263f919bb1988a27cbf3dbcdafb6e185dcfc9494aa596f91194bf6be96775" name="python-futures" arch="src">
  <version epoch="0" ver="3.0.5" rel="1.el7_9"/>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 2.1.6-1" date="1390219200">- 2.1.6</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.1.6-2" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Toshio Kuratomi &lt;toshio@fedoraproject.org&gt; - 2.1.6-3" date="1404129600">- Replace python-setuptools-devel BR with python-setuptools</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 3.0.2-1" date="1433678400">- 3.0.2</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.0.2-2" date="1434628800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 3.0.3-1" date="1435147200">- 3.0.3</changelog>
  <changelog author="Terje Rosten &lt;terje.rosten@ntnu.no&gt; - 3.0.4-1" date="1453118400">- 3.0.4</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 3.0.4-2" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.0.4-3" date="1468929600">- https://fedoraproject.org/wiki/Changes/Automatic_Provides_for_Python_RPM_Packages</changelog>
  <changelog author="Orion Poplawski &lt;orion@cora.nwra.com&gt; - 3.0.5-1" date="1479729600">- Update to 3.0.5
- Ship python2-futures</changelog>
</package>
<package pkgid="2efef42f85275b9cd0a84b4fcea2d1e6946498690ddf12dcc801d694090fdf58" name="python-markdown" arch="src">
  <version epoch="0" ver="2.4.1" rel="4.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.2.0-2" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="David Malcolm &lt;dmalcolm@redhat.com&gt; - 2.2.0-3" date="1344081600">- rebuild for https://fedoraproject.org/wiki/Features/Python_3.3</changelog>
  <changelog author="Thomas Moschny &lt;thomas.moschny@gmx.de&gt; - 2.2.1-1" date="1353672000">- Update to 2.2.1.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.2.1-2" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Dennis Gilmore &lt;dennis@ausil.us&gt; - 2.2.1-3" date="1389268800">- no python3 in RHEL 7</changelog>
  <changelog author="Thomas Moschny &lt;thomas.moschny@gmx.de&gt; - 2.4-1" date="1397563200">- Update to 2.4.
- Spec file cleanups.</changelog>
  <changelog author="Thomas Moschny &lt;thomas.moschny@gmx.de&gt; - 2.4.1-1" date="1401883200">- Update to 2.4.1.</changelog>
  <changelog author="Thomas Moschny &lt;thomas.moschny@gmx.de&gt; - 2.4.1-2" date="1504094400">- Enable python3 subpackage on EPEL7.</changelog>
  <changelog author="Orion Poplawski &lt;orion@nwra.com&gt; - 2.4.1-3" date="1543060800">- Ship python2-markdown and python3_other-markdown
- Modernize spec</changelog>
  <changelog author="Troy Dawson &lt;tdawson@redhat.com&gt; - 2.4.1-4" date="1552046400">- Rebuilt to change main python from 3.4 to 3.6</changelog>
</package>
<package pkgid="43d648fb2d4cc848cd2153551923c3e8c630cf19d2529f47708be8ac09492153" name="python-progressbar" arch="src">
  <version epoch="0" ver="2.3" rel="4.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.2-7" date="1248609600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="David Malcolm &lt;dmalcolm@redhat.com&gt; - 2.2-8" date="1279800000">- Rebuilt for https://fedoraproject.org/wiki/Features/Python_2.7/MassRebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.2-9" date="1297166400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.2-10" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.2-11" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.2-12" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Christof Damian &lt;christof@damian.net&gt; - 2.3-1" date="1367064000">- upstream 2.3</changelog>
  <changelog author="Christof Damian &lt;christof@damian.net&gt; - 2.3-2" date="1369483200">- added interrupt patch bug #965919</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.3-3" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.3-4" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
</package>
<package pkgid="89e618c1a8674c330c764d4a5610f1b38be2abeb2bf25a1160680989789716b4" name="python-psutil" arch="src">
  <version epoch="0" ver="2.2.1" rel="2.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.4.1-2" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Mohamed El Morabity &lt;melmorabity@fedoraproject.org&gt; - 0.5.1-1" date="1341144000">- Update to 0.5.1</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.5.1-2" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="David Malcolm &lt;dmalcolm@redhat.com&gt; - 0.5.1-3" date="1344081600">- rebuild for https://fedoraproject.org/wiki/Features/Python_3.3</changelog>
  <changelog author="Mohamed El Morabity &lt;melmorabity@fedorapeople.org&gt; - 0.6.1-1" date="1345377600">- Update to 0.6.1</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.6.1-2" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Matthias Runge &lt;mrunge@redhat.com&gt; - 0.6.1-3" date="1392724800">- epel doesn't have python3</changelog>
  <changelog author="Alan Pevec &lt;apevec@redhat.com&gt; - 1.2.1-1" date="1426075200">- Update to 1.2.1</changelog>
  <changelog author="Ralph Bean &lt;rbean@redhat.com&gt; - 2.2.1-1" date="1449662400">- Update to 2.2.1 for https://bugzilla.redhat.com/1288221
- Update names of %doc files.</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;besser82@fedoraproject.org&gt; - 2.2.1-2" date="1504180800">- Build a package for Python34</changelog>
</package>
<package pkgid="e46958989c801db1895dd624d9e49060b54ad3d714c3a1e6fb05b773a001df6b" name="python-pymongo" arch="src">
  <version epoch="0" ver="2.5.2" rel="4.el7_9"/>
  <changelog author="Andrew McNabb &lt;amcnabb@mcnabbs.org&gt; - 2.5-4" date="1366804800">- Fix the test running procedure</changelog>
  <changelog author="Andrew McNabb &lt;amcnabb@mcnabbs.org&gt; - 2.5-5" date="1370952000">- Bump the obsoletes version</changelog>
  <changelog author="Andrew McNabb &lt;amcnabb@mcnabbs.org&gt; - 2.5.2-1" date="1371038400">- Update to pymongo 2.5.2</changelog>
  <changelog author="Andrew McNabb &lt;amcnabb@mcnabbs.org&gt; - 2.5.2-2" date="1371124800">- Bump the obsoletes version for pymongo-gridfs</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.5.2-3" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Bohuslav Kabrda &lt;bkabrda@redhat.com&gt; - 2.5.2-4" date="1400068800">- Rebuilt for https://fedoraproject.org/wiki/Changes/Python_3.4</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.5.2-5" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.5.2-6" date="1408276800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.5.2-7" date="1434628800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Randy Barlow &lt;rbarlow@redhat.com&gt; - 2.5.2-4" date="1454414400">- Patch to use ssl_match_hostname from backports for CVE-2013-2099 and CVE-2013-7440.</changelog>
</package>
<package pkgid="64d87df529c0cfd0de78ba21b93fa58d4acb61fe294bc63ca29565a006aeb750" name="python-pyrfc3339" arch="src">
  <version epoch="0" ver="1.0" rel="2.el7_9"/>
  <changelog author="Felix Schwarz &lt;fschwarz@fedoraproject.org&gt; - 0.2-1" date="1446033600">- initial packaging</changelog>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt;    - 0.2-2" date="1446984000">- Update to follow the python guidelines</changelog>
  <changelog author="James Hogarth &lt;james.hogarth@gmail.com&gt;    - 1.0-1" date="1447156800">- Add installed tests back as per review
- Update to new 1.0 PyPi release
- Add external license file</changelog>
  <changelog author="Robert Buchholz &lt;rbu@goodpoint.de&gt; - 1.0-2" date="1449144000">- epel7: Only build python2 package</changelog>
</package>
<package pkgid="65aee9f0a2541d1bef8183f7e6f404a85336b9a46ff62af56f660f13827a601c" name="python-repoze-sphinx-autointerface" arch="src">
  <version epoch="0" ver="0.7.1" rel="6.el7_9"/>
  <changelog author="Jerry James &lt;loganjerry@gmail.com&gt; - 0.7.1-1" date="1401364800">- Initial RPM (bz 1102858)</changelog>
  <changelog author="Jerry James &lt;loganjerry@gmail.com&gt; - 0.7.1-2" date="1401710400">- Conditionalize the python3 package</changelog>
  <changelog author="Jerry James &lt;loganjerry@gmail.com&gt; - 0.7.1-3" date="1401796800">- Do not use the py3dir macro; see https://fedorahosted.org/fpc/ticket/435
- Remove the %clean script</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.7.1-4" date="1434628800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.7.1-5" date="1447156800">- Rebuilt for https://fedoraproject.org/wiki/Changes/python3.5</changelog>
  <changelog author="Nick Le Mouton &lt;nick@noodles.net.nz&gt; - 0.7.1-6" date="1450094400">- Cleaned up spec file
- Added el6 macros/requires</changelog>
</package>
<package pkgid="352f5660c096a8b2f0fa18e0781154e0267a0a2bb848e2f9784c60abc103f17c" name="python-teapot" arch="src">
  <version epoch="0" ver="2.6" rel="1.el7_9"/>
</package>
<package pkgid="d53e52aa6d65521727a54cd5688b50dd0be044880c2319104cacb007d5c840b1" name="python-zope-component" arch="src">
  <version epoch="1" ver="4.1.0" rel="5.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 4.1.0-2" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Ralph Bean &lt;rbean@redhat.com&gt; - 4.2.0-1" date="1392206400">- Latest upstream.</changelog>
  <changelog author="Bohuslav Kabrda &lt;bkabrda@redhat.com&gt; - 4.2.0-2" date="1400068800">- Rebuilt for https://fedoraproject.org/wiki/Changes/Python_3.4</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 4.2.0-3" date="1402228800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Ralph Bean &lt;rbean@redhat.com&gt; - 4.2.1-1" date="1405944000">- Latest upstream.</changelog>
  <changelog author="Aurelien Bompard &lt;abompard@fedoraproject.org&gt; - 1:4.1.0-1" date="1415361600">- Revert to version 4.1.0, EPEL7 can't have any later version because they
  require zope.interface &gt;= 4.1.0, which is in RHEL.
  See https://bugzilla.redhat.com/show_bug.cgi?id=1161506</changelog>
  <changelog author="Aurelien Bompard &lt;abompard@fedoraproject.org&gt; - 1:4.1.0-2" date="1493035200">- Enable creation of the Python 3 packages in EPEL, now that the dependencies
  are available.</changelog>
  <changelog author="Aurelien Bompard &lt;abompard@fedoraproject.org&gt; - 1:4.1.0-3" date="1493121600">- Use the included documentation.
- Simplify the building of the Python2 version.</changelog>
  <changelog author="Aurelien Bompard &lt;abompard@fedoraproject.org&gt; - 1:4.1.0-4" date="1501675200">- Fix BuildRequires name</changelog>
  <changelog author="Troy Dawson &lt;tdawson@redhat.com&gt; - 1:4.1.0-5" date="1552046400">- Rebuilt to change main python from 3.4 to 3.6</changelog>
</package>
<package pkgid="aab9e6482dcc81514a4400d033d9b823bc626161af669a2c518acdaf75c22521" name="python2-numpy" arch="src">
  <version epoch="1" ver="1.7.1" rel="0.el7_9"/>
  <changelog author="Jason L Tibbitts III &lt;tibbs@math.uh.edu&gt; - 1:1.7.1-0" date="1522152000">- Initial version.</changelog>
</package>
<package pkgid="ab408b6c824d70d5ce7f1546d0bd27e5e2097183892bd43dc7de66fb937e2396" name="python2-requests" arch="src">
  <version epoch="0" ver="2.6.0" rel="0.el7_9"/>
  <changelog author="Jason L Tibbitts III &lt;tibbs@math.uh.edu&gt; - 2.6.0-0" date="1518696000">- Initial version.</changelog>
</package>
<package pkgid="456235408d841e5dab92bd5d3d6e0e2de9c318931b4fbbb74d41cd474c286578" name="python3-nose" arch="src">
  <version epoch="0" ver="1.3.7" rel="4.el7_9"/>
  <changelog author="Orion Poplawski &lt;orion@cora.nwra.com&gt; - 1.3.7-1" date="1451476800">- Initial EPEL7 package</changelog>
  <changelog author="Orion Poplawski &lt;orion@cora.nwra.com&gt; - 1.3.7-2" date="1454414400">- Fix URL
- Fix long line in description
- Include more documentation</changelog>
  <changelog author="Carl George &lt;carl@george.computer&gt; - 1.3.7-3" date="1531483200">- Enable python36 subpackage
- Add patches 1, 2, 3, and 4 from Fedora for py36 compatibility</changelog>
  <changelog author="Troy Dawson &lt;tdawson@redhat.com&gt;" date="1551960000">- Rebuilt to change main python from 3.4 to 3.6</changelog>
</package>
<package pkgid="510bcf92bd0ace6a96111462dbad004b7528815f0a2efc53670ebbd02533939d" name="radcli" arch="src">
  <version epoch="0" ver="1.2.11" rel="3.el7_9"/>
  <changelog author="Nikos Mavrogiannopoulos &lt;nmav@redhat.com&gt; - 1.2.8-1" date="1498824000">- New upstream release</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.8-2" date="1501156800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.8-3" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Nikos Mavrogiannopoulos &lt;nmav@redhat.com&gt; - 1.2.9-1" date="1516622400">- New upstream release</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.9-2" date="1518177600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Nikos Mavrogiannopoulos &lt;nmav@redhat.com&gt; - 1.2.10-1" date="1526385600">- New upstream release</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.10-2" date="1531569600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Nikos Mavrogiannopoulos &lt;nmav@redhat.com&gt; - 1.2.11-1" date="1536926400">- New upstream release</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.11-2" date="1549108800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.2.11-3" date="1564142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
</package>
<package pkgid="e95c8551b621c4bbfb24caf855845000e3f888757bf7258639ff4847cce78230" name="radiusclient-ng" arch="src">
  <version epoch="0" ver="0.5.6" rel="5.el7_9"/>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.5.2-1" date="1148904000">- First version for Fedora Extras</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.5.2-2" date="1152619200">- Add BR so that building in minimal buildroot works
- Run libtoolize.</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.5.2-3" date="1152619201">- Own /etc/radiusclient-ng</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.5.2-4" date="1156766400">- Bump release and rebuild.</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.5.5.1-1" date="1180094400">- Update to 0.5.5.1</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.5.6-2" date="1187179200">- Update to 0.5.6</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.5.6-3" date="1201953600">- Update patch to properly perform substitutions. (BZ#236350)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.5.6-4" date="1235563200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.5.6-5" date="1248609600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
</package>
<package pkgid="e2c08fef070927eb3ca6796e0a40d7eb375d6829eac905b84d7aa3f5fa10f88e" name="rhash" arch="src">
  <version epoch="0" ver="1.3.4" rel="2.el7_9"/>
  <changelog author="Orion Poplawski &lt;orion@cora.nwra.com&gt; - 1.3.4-1" date="1489060800">- Initial Fedora package</changelog>
  <changelog author="Orion Poplawski &lt;orion@cora.nwra.com&gt; - 1.3.4-2" date="1490011200">- Add %check section</changelog>
</package>
<package pkgid="6a115721459cc8b557e199f09a063d16eb84cf2451598e564f350b0b7ad3f5d8" name="ruby-augeas" arch="src">
  <version epoch="0" ver="0.5.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.2.0-3" date="1248609600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="David Lutterkort &lt;lutter@redhat.com&gt; - 0.3.0-1" date="1250078400">- New version</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.3.0-2" date="1297252800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="David Lutterkort &lt;lutter@redhat.com&gt; - 0.4.0-1" date="1301400000">- Require augeas-0.8.0</changelog>
  <changelog author="David Lutterkort &lt;lutter@redhat.com&gt; - 0.4.1-1" date="1301400001">- New version</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.4.1-2" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Bohuslav Kabrda &lt;bkabrda@redhat.com&gt; - 0.4.1-3" date="1328616000">- Rebuilt for Ruby 1.9.3.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.4.1-4" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.4.1-5" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="David Lutterkort &lt;lutter@redhat.com&gt; - 0.5.0-1" date="1363176000">- New version; updated spec file for latest guidelines</changelog>
</package>
<package pkgid="bcd808435357d5e88c14db6c2f787bc0de0545dd36c3b0ed9ab6a6522afd1c41" name="ruby-shadow" arch="src">
  <version epoch="0" ver="1.4.1" rel="23.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4.1-14" date="1297252800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4.1-15" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Todd Zullinger &lt;tmz@pobox.com&gt; - 1.4.1-16" date="1334923200">- Add ruby-1.9 support</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4.1-17" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4.1-18" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 1.4.1-19" date="1363608000">- Rebuild for https://fedoraproject.org/wiki/Features/Ruby_2.0.0</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.4.1-20" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Orion Poplawski &lt;orion@cora.nwra.com&gt; - 1.4.1-21" date="1390478400">- Update ruby requires conditionals for EPEL7</changelog>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 1.4.1-22" date="1398686400">- Rebuilt for https://fedoraproject.org/wiki/Changes/Ruby_2.1</changelog>
  <changelog author="Sam Kottler &lt;skottler@fedoraproject.org&gt; - 1.4.1-23" date="1401537600">- Fix conditional so shadow.so gets installed to the proper location (BZ#1094046)</changelog>
</package>
<package pkgid="5b0a404b5beda9dddb78f2c4b61127030654a4eeef6e25803259290f0659d1cf" name="rubygem-diff-lcs" arch="src">
  <version epoch="0" ver="1.1.3" rel="2.2.el7_9"/>
  <changelog author="Lubomir Rintel (Good Data) &lt;lubo.rintel@gooddata.com&gt; - 1.1.2-2" date="1244462400">- Depend on ruby(abi)
- Replace defines with globals</changelog>
  <changelog author="Lubomir Rintel (Good Data) &lt;lubo.rintel@gooddata.com&gt; - 1.1.2-3" date="1246017600">- Get rid of duplicate files (thanks to Mamoru Tasaka)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.2-4" date="1248609600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.2-5" date="1297252800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.2-6" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Mamoru Tasaka &lt;mtasaka@fedoraproject.org&gt; - 1.1.2-7" date="1327233600">- Rebuild against ruby 1.9</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.2-8" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 1.1.3-1" date="1345204800">- Update to diff-lcs 1.1.3.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.3-2" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 1.1.3-3" date="1361361600">- Rebuild for https://fedoraproject.org/wiki/Features/Ruby_2.0.0
- Change the dependency to rubygem(rspec).
- Add bootstrap code.</changelog>
</package>
<package pkgid="c62d5e186bbd38b112ee86d187e1bc694fdc584c609b0f9ff6fce405c81ab110" name="rubygem-fakefs" arch="src">
  <version epoch="0" ver="0.13.1" rel="3.el7_9"/>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 0.6.7-1" date="1435147200">- Update to FakeFS 0.6.7.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.6.7-2" date="1454587200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 0.10.0-1" date="1478865600">- Update to FakeFS 0.10.0.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.10.0-2" date="1486814400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.10.0-3" date="1501156800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 0.11.1-1" date="1503489600">- Update to FakeFS 0.11.1.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.11.1-2" date="1518177600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 0.13.1-1" date="1518523200">- Update to FakeFS 0.13.1.</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.13.1-2" date="1531569600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Ha&#xEF;kel Gu&#xE9;mar &lt;hguemar@fedoraproject.org&gt; - 0.13.1-3" date="1541678400">- Fix EL7 build</changelog>
</package>
<package pkgid="c7bb4cff84d0862563a92066beea0b08473484f5263ce9394ec39f29b0f4b3fc" name="rubygem-nokogiri" arch="src">
  <version epoch="0" ver="1.6.1" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.5.5-1.1" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 1.5.5-2" date="1345204800">- Rebuilt againts libxml2 2.9.</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 1.5.6-1" date="1357041600">- A Happy New Year
- 1.5.6</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.5.6-1.2" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 1.5.6-2" date="1361620800">- Rebuild for https://fedoraproject.org/wiki/Features/Ruby_2.0.0</changelog>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 1.5.6-3" date="1364299200">- Use %{gem_extdir_mri} instead of %{gem_extdir}.</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 1.5.9-1" date="1364472000">- 1.5.9</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.5.9-1.1" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 1.6.0-1" date="1380888000">- 1.6.0</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 1.6.1-1" date="1387972800">- 1.6.1</changelog>
</package>
<package pkgid="b08be2b3577f3b5706e38c3b781e1cafe8e08d957fb14ae78d57fdd7d137d2d4" name="rubygem-pathspec" arch="src">
  <version epoch="0" ver="0.2.1" rel="3.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.0.2-5" date="1486814400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Athos Ribeiro &lt;athoscr@fedoraproject.org&gt; - 0.1.0-1" date="1496664000">- New version</changelog>
  <changelog author="Athos Ribeiro &lt;athoscr@fedoraproject.org&gt; - 0.1.0-2" date="1496664001">- Remove unused Source</changelog>
  <changelog author="Athos Ribeiro &lt;athoscr@fedoraproject.org&gt; - 0.1.0-3" date="1496664002">- Do not use simplecov
- Add spec directory to documentation subpackage
- Remove code for EOL fedora versions</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.0-4" date="1501156800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Athos Ribeiro &lt;athoscr@fedoraproject.org&gt; - 0.1.1-1" date="1504699200">- Update version</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.1.1-2" date="1518177600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Athos Ribeiro &lt;athoscr@fedoraproject.org&gt; - 0.2.1-1" date="1518523200">- Update version</changelog>
  <changelog author="Athos Ribeiro &lt;athoscr@fedoraproject.org&gt; - 0.2.1-2" date="1518523201">- Bump release
- Ship new binary file</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.2.1-3" date="1531569600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
</package>
<package pkgid="9c5307a4761f263723ad421c934fac5e7881410f0b10b5f22ee2a6f5df1a1924" name="rubygem-pathutil" arch="src">
  <version epoch="0" ver="0.14.0" rel="1.el7_9"/>
  <changelog author="Bj&#xF6;rn Esser &lt;fedora@besser82.io&gt; - 0.14.0-0.1" date="1471780800">- initial rpm-release (#1368849)</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;fedora@besser82.io&gt; - 0.14.0-1" date="1471953600">- initial import (#1368849)</changelog>
</package>
<package pkgid="4c5a78a377fd868026aaed1d96b7762da6bf31cea4cb42ce337485fa45aa636c" name="rubygem-rgen" arch="src">
  <version epoch="0" ver="0.6.6" rel="2.el7_9"/>
  <changelog author="Sam Kottler &lt;shk@redhat.com&gt; - 0.6.6-1" date="1389009600">- Initial package</changelog>
  <changelog author="Sam Kottler &lt;shk@redhat.com&gt; - 0.6.6-2" date="1390392000">- Fixes based on review feedback</changelog>
</package>
<package pkgid="5a05ebc0742f4c1effdd0c23a772d93678f1ca845bc1cd79021f99bc2ea1ca99" name="rubygem-rspec" arch="src">
  <version epoch="0" ver="2.14.1" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.3.1-2" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="V&#xED;t Ondruch &lt;bkabrda@redhat.com&gt; - 2.8.0-1" date="1330948800">- Update to RSpec 2.8.0.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.8.0-2" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Mamoru Tasaka &lt;mtasaka@fedoraproject.org&gt; - 2.11.0-1" date="1349956800">- Update to Rspec 2.11.0</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.12.0-1" date="1357128000">- Update to Rspec 2.12.0</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.12.0-2" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 2.12.0-3" date="1361361600">- Rebuild for https://fedoraproject.org/wiki/Features/Ruby_2.0.0</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.13.0-1" date="1364472000">- 2.13.0</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.13.0-2" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.og&gt; - 2.14.1-1" date="1376654400">- 2.14.1</changelog>
</package>
<package pkgid="8d684553a1bc226b78da34ee193d94653f3fb40c16362fd04da00a7413b4c732" name="rubygem-rspec-core" arch="src">
  <version epoch="0" ver="2.14.8" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.13.1-2.1" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.13.1-3" date="1375790400">- Bootstrap for rubygem-gherkin &lt;- rubygem-cucumber</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt;" date="1375790401">- Again enable test suite</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.5-1" date="1376654400">- 2.14.5</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.5-2" date="1376654401">- Enable test suite again</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.6-1" date="1382616000">- 2.14.6</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.7-1" date="1384171200">- 2.14.7</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.8-1" date="1394107200">- 2.14.8</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.14.8-1.1" date="1402228800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Troy Dawson &lt;tdawson@redhat.com&gt; - 2.14.8-1.0" date="1406894400">- Remove testing to break dependency circle</changelog>
</package>
<package pkgid="6f346fd45e11083dc23d39eaf7ac938337ae48290e541e34a6f23cadf44b85ca" name="rubygem-rspec-expectations" arch="src">
  <version epoch="0" ver="2.14.5" rel="2.el7_9"/>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.13.0-1" date="1364472000">- 2.13.0</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.13.0-2" date="1364472001">- Enable test suite again</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.13.0-2.1" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.2-1" date="1376654400">- 2.14.2</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.2-2" date="1376654401">- Enable test suite again</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.3-1" date="1380283200">- 2.14.3</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.4-1" date="1384171200">- 2.14.4</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.5-1" date="1391428800">- 2.14.5</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.14.5-1.1" date="1402228800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Bryan Kearney &lt;bkearney@redhat.com&gt; - 2.14.5-2" date="1403006400">- Turning off test to import into epel7</changelog>
</package>
<package pkgid="6dee668deb12205c791689f347b0d1da67fbb6633fc88ca3994bdf65b2e4a294" name="rubygem-rspec-mocks" arch="src">
  <version epoch="0" ver="2.14.6" rel="2.el7_9"/>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.13.0-2" date="1364472000">- Enable test suite again</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.13.1-1" date="1365768000">- 2.13.1</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.13.1-1.1" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.3-1" date="1376654400">- 2.14.3</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.3-2" date="1376654401">- Enable test suite again</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.4-1" date="1382616000">- 2.14.4</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.5-1" date="1391515200">- 2.14.5</changelog>
  <changelog author="Mamoru TASAKA &lt;mtasaka@fedoraproject.org&gt; - 2.14.6-1" date="1393502400">- 2.14.6</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.14.6-1.1" date="1402228800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Bryan Kearney &lt;bkearney@redhat.com&gt; - 2.14.6-2" date="1403006400">- Turning off test to import into epel7</changelog>
</package>
<package pkgid="6f31960e7c8c5bb76b3a837d1d99ad703a54e650e8b3d985d7fa738f6bf52954" name="schroedinger" arch="src">
  <version epoch="0" ver="1.0.11" rel="4.el7_9"/>
  <changelog author="Nicolas CHauvet &lt;kwizart@fedoraproject.org&gt; - 1.0.8-4" date="1265025600">- Remove gstreamer-plugins-schroedinger 
  Obsoleted by gst-plugins-bad-free introduction in Fedora.</changelog>
  <changelog author="Fabian Deutsch &lt;fabiand@fedoraproject.org&gt; - 1.0.9-1" date="1267790400">- Update to 1.0.9
- Dropped dependency on liboil
- Added dependency on orc</changelog>
  <changelog author="Fabian Deutsch &lt;fabiand@fedoraproject.org&gt; - 1.0.9-2" date="1271937600">- Added dependency on gtk-doc</changelog>
  <changelog author="Fabian Deutsch &lt;fabiand@fedoraproject.org&gt; - 1.0.10-1" date="1287921600">- Update to 1.0.10</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0.10-2" date="1297252800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0.10-3" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fabian Deutsch &lt;fabiand@fedoraproject.org&gt; - 1.0.11-1" date="1327320000">- Update to 1.0.11</changelog>
  <changelog author="Fabian Deutsch &lt;fabiand@fedoraproject.org&gt; - 1.0.11-2" date="1327665600">- Bump required orc version to 0.4.16</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0.11-3" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0.11-4" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
</package>
<package pkgid="81180bb2a52eb99468c4900ae9228d4a4cbcfd07c03866e2badacfca5ac86d45" name="scons" arch="src">
  <version epoch="0" ver="3.0.1" rel="2.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.2.0-2" date="1360843200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Fabian Affolter &lt;mail@fabian-affolter.ch&gt; - 2.3.0-1" date="1362830400">- Updated to new upstream version 2.3.0</changelog>
  <changelog author="Honza Horak &lt;hhorak@redhat.com&gt; - 2.3.0-1" date="1367755200">- Add support for software collections</changelog>
  <changelog author="Honza Horak &lt;hhorak@redhat.com&gt; - 2.3.4-1" date="1421668800">- Update to new upstream version 2.3.4</changelog>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 2.3.4-2" date="1421841600">- Allow to use python from SCL</changelog>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 2.3.4-3" date="1426680000">- Rebuilt for 'scls' removal</changelog>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 2.5.1-1" date="1496750400">- Rebased to latest version from Fedora 27</changelog>
  <changelog author="Marek Skalicky &lt;mskalick@redhat.com&gt; - 2.5.1-3" date="1496750401">- Relase bump to be able to built scons for RHSCL 3.0</changelog>
  <changelog author="Marek Skalick&#xFD; &lt;mskalick@redhat.com&gt; - 2.5.1-4" date="1497528000">- Requires python 2.x, but on rhel6 it is not labeled as python2. So don't use
  python2.</changelog>
  <changelog author="Marek Skalick&#xFD; &lt;mskalick@redhat.com&gt; - 3.0.1-2" date="1513339200">- Rebase scons 3 from Fedora rawhide</changelog>
</package>
<package pkgid="2d747e475bb95f131ef7558a4abd183c5b7705b593ab953bde92a78ac4da1347" name="serd" arch="src">
  <version epoch="0" ver="0.30.2" rel="2.el7_9"/>
  <changelog author="Guido Aulisi &lt;guido.aulisi@gmail.com&gt; - 0.28.0-1" date="1502366400">- Update to 0.28.0
- Use license macro</changelog>
  <changelog author="Kalev Lember &lt;klember@redhat.com&gt; - 0.28.0-2" date="1502452800">- Bump and rebuild for an rpm signing issue</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.28.0-3" date="1518177600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.28.0-4" date="1531569600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Guido Aulisi &lt;guido.aulisi@gmail.com&gt; - 0.28.0-5" date="1531656000">- Fix FTBFS due to the move of /usr/bin/python into a separate package</changelog>
  <changelog author="Guido Aulisi &lt;guido.aulisi@gmail.com&gt; - 0.30.0-1" date="1532606400">- Update to 0.30.0
- Remove ldconfig scriptlets
- Minor spec cleanup</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.30.0-2" date="1549108800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.30.0-3" date="1564142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Guido Aulisi &lt;guido.aulisi@gmail.com&gt; - 0.30.2-1" date="1575633600">- Update to 0.30.2
- Use python3</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.30.2-2" date="1580385600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild</changelog>
</package>
<package pkgid="469628d5cb2bcd3f91fb7911d79c48c03c1d209eda1c3cc138988c015e027438" name="sipp" arch="src">
  <version epoch="0" ver="3.5.1" rel="1.el7_9"/>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 3.2-3" date="1320321600">- Fix authorization
- Cherry-picked two patches from svn trunk</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2-4" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2-5" date="1330430400">- Rebuilt for c++ ABI breakage</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2-6" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2-7" date="1360929600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 3.3-1" date="1360929601">- Ver. 3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.4.0-1" date="1435752000">- Updated to version 3.4.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.4.1-1" date="1435752001">- Updated to version 3.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.5.0-1" date="1477915200">- Updated to version 3.5.0
- Added gsl support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.5.1-1" date="1478001600">- Updated to version 3.5.1</changelog>
</package>
<package pkgid="9ca522b09b5b305239e13bf794af5694e6d912d5a1e4e61172fb67f5f1366426" name="sipp" arch="src">
  <version epoch="0" ver="3.5.2" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2-4" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2-5" date="1330430400">- Rebuilt for c++ ABI breakage</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2-6" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2-7" date="1360929600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 3.3-1" date="1360929601">- Ver. 3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.4.0-1" date="1435752000">- Updated to version 3.4.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.4.1-1" date="1435752001">- Updated to version 3.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.5.0-1" date="1477915200">- Updated to version 3.5.0
- Added gsl support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.5.1-1" date="1478001600">- Updated to version 3.5.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.5.2-1" date="1531483200">- Updated to version 3.5.2</changelog>
</package>
<package pkgid="7df76fd99a2c2702e33d4f0d0d695790d671184724f94e202348ef7108978be3" name="sipp" arch="src">
  <version epoch="0" ver="3.5.3" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2-5" date="1330430400">- Rebuilt for c++ ABI breakage</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2-6" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2-7" date="1360929600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 3.3-1" date="1360929601">- Ver. 3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.4.0-1" date="1435752000">- Updated to version 3.4.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.4.1-1" date="1435752001">- Updated to version 3.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.5.0-1" date="1477915200">- Updated to version 3.5.0
- Added gsl support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.5.1-1" date="1478001600">- Updated to version 3.5.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.5.2-1" date="1531483200">- Updated to version 3.5.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.5.3-1" date="1560859200">- Updated to version 3.5.3</changelog>
</package>
<package pkgid="ad85591beef96ea28e6ebf36acc2ad080ceea23f5180b4e65ad229aaa35d81f4" name="sipp" arch="src">
  <version epoch="0" ver="3.6.0" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2-6" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 3.2-7" date="1360929600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Peter Lemenkov &lt;lemenkov@gmail.com&gt; - 3.3-1" date="1360929601">- Ver. 3.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.4.0-1" date="1435752000">- Updated to version 3.4.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.4.1-1" date="1435752001">- Updated to version 3.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.5.0-1" date="1477915200">- Updated to version 3.5.0
- Added gsl support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.5.1-1" date="1478001600">- Updated to version 3.5.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.5.2-1" date="1531483200">- Updated to version 3.5.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.5.3-1" date="1560859200">- Updated to version 3.5.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 3.6.0-1" date="1560859201">- Updated to version 3.6.0</changelog>
</package>
<package pkgid="2732a224bad2811a5e63e829974311f31aa04a0a3875cdb91bca94b71dd38586" name="sngrep" arch="src">
  <version epoch="0" ver="1.4.10" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1477915200">- Updated to release 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1482321600">- Updated to release 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1493208000">- Updated to release 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1528545600">- Updated to release 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1528545601">- Updated to release 1.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1540987200">- Updated to release 1.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1605009600">- Updated to release 1.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1605009601">- Updated to release 1.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1621512000">- Updated to release 1.4.9</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.10-1" date="1637582400">- Updated to release 1.4.10</changelog>
</package>
<package pkgid="52cd02338339ca726bab51c62d286bc53f8da8aff75df0520a3be2b41d7742c2" name="sngrep" arch="src">
  <version epoch="0" ver="1.4.3" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.4.2-1" date="1447848000">- Updated to release 0.4.2
- Enabled IPv6 support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.0.0-1" date="1447848001">- Updated to release 1.0.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.0-1" date="1447848002">- Updated to release 1.1.0
- Enabled EEP/HEP support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.0-1" date="1450699200">- Updated to release 1.2.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.0-1" date="1477915200">- Updated to release 1.3.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1477915201">- Updated to release 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.0-1" date="1477915202">- Updated to release 1.4.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1477915203">- Updated to release 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1482321600">- Updated to release 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1493208000">- Updated to release 1.4.3</changelog>
</package>
<package pkgid="351297914dcc1604af3f14768d40e26d26380946a4110731f12b9f5f1fd79507" name="sngrep" arch="src">
  <version epoch="0" ver="1.4.4" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.0.0-1" date="1447848000">- Updated to release 1.0.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.0-1" date="1447848001">- Updated to release 1.1.0
- Enabled EEP/HEP support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.0-1" date="1450699200">- Updated to release 1.2.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.0-1" date="1477915200">- Updated to release 1.3.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1477915201">- Updated to release 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.0-1" date="1477915202">- Updated to release 1.4.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1477915203">- Updated to release 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1482321600">- Updated to release 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1493208000">- Updated to release 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1528545600">- Updated to release 1.4.4</changelog>
</package>
<package pkgid="569cfe247ecc48b88bf6878bc74c147180c44f4b076af57b176d42d38086eec2" name="sngrep" arch="src">
  <version epoch="0" ver="1.4.5" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.1.0-1" date="1447848000">- Updated to release 1.1.0
- Enabled EEP/HEP support</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.0-1" date="1450699200">- Updated to release 1.2.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.0-1" date="1477915200">- Updated to release 1.3.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1477915201">- Updated to release 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.0-1" date="1477915202">- Updated to release 1.4.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1477915203">- Updated to release 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1482321600">- Updated to release 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1493208000">- Updated to release 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1528545600">- Updated to release 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.5-1" date="1528545601">- Updated to release 1.4.5</changelog>
</package>
<package pkgid="b66bf10de96c33dd7b1089770cb8447e34c4e0430ce68fdc7bf293ae63a2f50a" name="sngrep" arch="src">
  <version epoch="0" ver="1.4.6" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.2.0-1" date="1450699200">- Updated to release 1.2.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.0-1" date="1477915200">- Updated to release 1.3.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1477915201">- Updated to release 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.0-1" date="1477915202">- Updated to release 1.4.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1477915203">- Updated to release 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1482321600">- Updated to release 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1493208000">- Updated to release 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1528545600">- Updated to release 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1528545601">- Updated to release 1.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1540987200">- Updated to release 1.4.6</changelog>
</package>
<package pkgid="9ac734341b72fe04b201e8cf87fdb9bdd6828e4dbf664a25c38595d1a35ad57b" name="sngrep" arch="src">
  <version epoch="0" ver="1.4.7" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.0-1" date="1477915200">- Updated to release 1.3.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1477915201">- Updated to release 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.0-1" date="1477915202">- Updated to release 1.4.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1477915203">- Updated to release 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1482321600">- Updated to release 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1493208000">- Updated to release 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1528545600">- Updated to release 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1528545601">- Updated to release 1.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1540987200">- Updated to release 1.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1605009600">- Updated to release 1.4.7</changelog>
</package>
<package pkgid="a7c13f27510d765e0be062f29fe8aa077821f8d2f88d244b1dc692f6bfb499b5" name="sngrep" arch="src">
  <version epoch="0" ver="1.4.8" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.3.1-1" date="1477915200">- Updated to release 1.3.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.0-1" date="1477915201">- Updated to release 1.4.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1477915202">- Updated to release 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1482321600">- Updated to release 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1493208000">- Updated to release 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1528545600">- Updated to release 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1528545601">- Updated to release 1.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1540987200">- Updated to release 1.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1605009600">- Updated to release 1.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1605009601">- Updated to release 1.4.8</changelog>
</package>
<package pkgid="e667f0c1be31fcf454332298d0af117db9c8f82e22fea05e4006e98fe7e842c6" name="sngrep" arch="src">
  <version epoch="0" ver="1.4.9" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.0-1" date="1477915200">- Updated to release 1.4.0</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.1-1" date="1477915201">- Updated to release 1.4.1</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.2-1" date="1482321600">- Updated to release 1.4.2</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.3-1" date="1493208000">- Updated to release 1.4.3</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1528545600">- Updated to release 1.4.4</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.4-1" date="1528545601">- Updated to release 1.4.5</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.6-1" date="1540987200">- Updated to release 1.4.6</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.7-1" date="1605009600">- Updated to release 1.4.7</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.8-1" date="1605009601">- Updated to release 1.4.8</changelog>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.4.9-1" date="1621512000">- Updated to release 1.4.9</changelog>
</package>
<package pkgid="71fb51919846bc05b37494850988960140bb2370b67212755e9c76bd42d45dbe" name="sofia-sip" arch="src">
  <version epoch="0" ver="1.12.11" rel="20.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.12.11-11" date="1434715200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.12.11-12" date="1454673600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.12.11-13" date="1486814400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.12.11-14" date="1501156800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.12.11-15" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.12.11-16" date="1518177600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.12.11-17" date="1531569600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.12.11-18" date="1549195200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.12.11-19" date="1564142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.12.11-20" date="1580385600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild</changelog>
</package>
<package pkgid="e7dc9d516f4f64c4f6419fa6ee3c13dbd157b48c6f1e33bf50b2b974dfe71374" name="sord" arch="src">
  <version epoch="0" ver="0.16.4" rel="3.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.16.0-4" date="1518177600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.16.0-5" date="1531569600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_29_Mass_Rebuild</changelog>
  <changelog author="Guido Aulisi &lt;guido.aulisi@gmail.com&gt; - 0.16.0-6" date="1531656000">- Fix FTBFS due to the move of /usr/bin/python into a separate package</changelog>
  <changelog author="Guido Aulisi &lt;guido.aulisi@gmail.com&gt; - 0.16.2-1" date="1532952000">- Update to 0.16.2
- Remove ldconfig scriptlets
- Minor spec cleanup</changelog>
  <changelog author="Florian Weimer &lt;fweimer@redhat.com&gt; - 0.16.2-2" date="1533038400">- Rebuild with fixed binutils</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.16.2-3" date="1549195200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_30_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.16.2-4" date="1564142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_31_Mass_Rebuild</changelog>
  <changelog author="Guido Aulisi &lt;guido.aulisi@gmail.com&gt; - 0.16.4-1" date="1575633600">- Update to 0.16.4
- Use python3</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 0.16.4-2" date="1580385600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_32_Mass_Rebuild</changelog>
  <changelog author="Guido Aulisi &lt;guido.aulisi@gmail.com&gt; - 0.16.4-3" date="1584273600">- Rebuilt for possible GCC 10 bug on power64 and arm</changelog>
</package>
<package pkgid="a040cb16dbb7d9df9d95fc0f41aa3260c17ed12fac38940642583f41f3486f6f" name="soxr" arch="src">
  <version epoch="0" ver="0.1.2" rel="1.el7_9"/>
  <changelog author="Richard Shaw &lt;hobbes1069@gmail.com&gt; - 0.1.1-1" date="1367236800">- Initial packaging.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.1.1-2" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.1.1-3" date="1402228800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.1.1-4" date="1408363200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.1.1-5" date="1434715200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Richard Shaw &lt;hobbes1069@gmail.com&gt; - 0.1.2-1" date="1441713600">- Update to latest upstream release.</changelog>
</package>
<package pkgid="27bb4a4d4e352887c5a700e737eaa7e370983203d2c46057db08acb59e92e131" name="spandsp" arch="src">
  <version epoch="0" ver="0.0.6" rel="0.8.pre21.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.0.5-0.2.pre4" date="1235563200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.0.5-0.3.pre4" date="1248609600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.0.6-0.1.pre12" date="1248782400">- Update to 0.0.6pre12</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.0.6-0.2.pre17" date="1280664000">- Update to 0.0.6pre17</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.0.6-0.3.pre17" date="1297252800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Jeffrey C. Ollie &lt;jeff@ocjtech.us&gt; - 0.0.6-0.4.pre18" date="1297252801">- 0.0.6pre18</changelog>
  <changelog author="Rex Dieter &lt;rdieter@fedoraproject.org&gt; 0.0.6-0.5.pre18" date="1324987200">- use of brackets confuses autotools (#691039)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.0.6-0.6.pre18" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.0.6-0.7.pre18" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Jeffrey Ollie &lt;jeff@ocjtech.us&gt; - 0.0.6-0.8.pre21" date="1350561600">- Update to 0.0.6pre21</changelog>
</package>
<package pkgid="db214d4d052efff245c12a0d6edfac0b49974e0d9ac2a54c84ca46d5aab45bb4" name="sshuttle" arch="src">
  <version epoch="0" ver="0" rel="8.20120810git9ce2fa0.el7_9"/>
  <changelog author="pascal.bleser@opensuse.org" date="1304510400">- initial version (0.20110503)</changelog>
  <changelog author="Marcel Wysocki &lt;maci@satgnu.net&gt; - 0.20121018-3" date="1350561600">- update to newer spapshot
- fedora port
- adhere more to packaging guidelines</changelog>
  <changelog author="Marcel Wysocki &lt;maci@satgnu.net&gt; - 20121019-1.gitg9ce2fa0" date="1350648000">- update to newer spapshot
- adhere more to packaging guidelines</changelog>
  <changelog author="Marcel Wysocki &lt;maci@satgnu.net&gt; - 20121019-2.gitg9ce2fa0" date="1350648001">- use .tar.gz instead of .zip</changelog>
  <changelog author="Marcel Wysocki &lt;maci@satgnu.net&gt; - 20121019-3.gitg9ce2fa0" date="1350907200">- Don't use macros for rm, install, chmod and cp.
- Remove defattr, since we're not going for EPEL4.
- Remove the clean section since we're not going for EPEL5.</changelog>
  <changelog author="Marcel Wysocki &lt;maci@satgnu.net&gt; - 20121019-4.gitg9ce2fa0" date="1350907201">- add missing dependencies</changelog>
  <changelog author="Marcel Wysocki &lt;maci@satgnu.net&gt; - 20121019-5.gitg9ce2fa0" date="1351684800">- remove python from deps and builddeps, rpm picks it up automatically 
- add manual page
- add missing builddep for manual generation</changelog>
  <changelog author="Marcel Wysocki &lt;maci@satgnu.net&gt; - 0-6.20120810git9ce2fa0" date="1353672000">- fixed hash ( woops, where did that g come from )
- fixed date to be commit date and not clone date
- use version 0 
- use datadir instead of /usr/local
- remove sshuttle.md in favor of the manpage
- remove make from BR
- added comment on how to create the source tarball</changelog>
  <changelog author="Marcel Wysocki &lt;maci@satgnu.net&gt; - 0-7.20120810git9ce2fa0" date="1357560000">- don't use doc macro for files in mandir</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0-8.20120810git9ce2fa0" date="1360929600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
</package>
<package pkgid="fb27211bcf10f70c5a08393bbafb4dd3c0d1ad3b25ace798d131f2a9a3cd0513" name="strongswan" arch="src">
  <version epoch="0" ver="5.7.2" rel="1.el7_9"/>
  <changelog author="Paul Wouters &lt;pwouters@redhat.com&gt; - 5.6.1-1" date="1513944000">- Updated to 5.6.1 (RSA-PSS support)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 5.6.1-2" date="1518177600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="Lubomir Rintel &lt;lkundrak@v3.sk&gt; - 5.6.2-1" date="1519214400">- Updated to 5.6.2 (Dropped libnm-glib use in charon-nm)</changelog>
  <changelog author="Bj&#xF6;rn Esser &lt;besser82@fedoraproject.org&gt; - 5.6.2-2" date="1520337600">- Rebuilt for libjson-c.so.4 (json-c v0.13.1)</changelog>
  <changelog author="Mikhail Zabaluev &lt;mikhail.zabaluev@gmail.com&gt; - 5.6.2-3" date="1526817600">- Move eap-radius, sqlite, and pkcs7 plugins out of tnc-imcvs, added package
  sqlite (#1579945)</changelog>
  <changelog author="Paul Wouters &lt;pwouters@redhat.com&gt; - 5.6.2-5" date="1527163200">- Resolves rhbz#1574939 IKEv2 VPN connections fail to use DNS servers provided by the server
- Resolves rhbz#1449875 Strongswan on epel built without the sql plugin but with the sqlite plugin</changelog>
  <changelog author="Paul Wouters &lt;pwouters@redhat.com&gt; - 5.6.2-6" date="1527163201">- Resolves rhbz#1581868 CVE-2018-5388 strongswan: buffer underflow in stroke_socket.c</changelog>
  <changelog author="Mikhail Zabaluev &lt;mikhail.zabaluev@gmail.com&gt; - 5.6.3-1" date="1527595200">- New version 5.6.3</changelog>
  <changelog author="Mikhail Zabaluev &lt;mikhail.zabaluev@gmail.com&gt; - 5.7.1-1" date="1538654400">- Updated to 5.7.1
- Resolves rhbz#1635872 CVE-2018-16152
- Resolves rhbz#1635875 CVE-2018-16151</changelog>
  <changelog author="Paul Wouters &lt;pwouters@redhat.com&gt; - 5.7.2-1" date="1547035200">- Updated to 5.7.2</changelog>
</package>
<package pkgid="5e8a05118c712f5a6e56a162aa9e088c7bcb257550c3ece05a478534e9690876" name="svgalib" arch="src">
  <version epoch="0" ver="1.9.25" rel="22.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.9.25-13" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Jaromir Capik &lt;jcapik@redhat.com&gt; - 1.9.25-14" date="1379592000">- Fixing segmentation faults in vga_getmodenumber (#787147)
- Cleaning the spec</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.9.25-15" date="1402228800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.9.25-16" date="1408363200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.9.25-17" date="1434715200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Jaromir Capik &lt;jcapik@redhat.com&gt; - 1.9.25-18" date="1438603200">- Fixing FTBFS caused by quickmath.h function redefinitions (#1240045)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.9.25-19" date="1454673600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.9.25-20" date="1486814400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.9.25-21" date="1501156800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.9.25-22" date="1501761600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Binutils_Mass_Rebuild</changelog>
</package>
<package pkgid="a7d95162c03106e8a877187be63c1fadfcd869aba1b281bed424670ba05d1803" name="tcpreplay" arch="src">
  <version epoch="0" ver="4.2.5" rel="1.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.1.0-3" date="1454673600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Bojan Smojver &lt;bojan@rexursive com&gt; - 4.1.1-1" date="1461931200">- bump up to 4.1.1</changelog>
  <changelog author="Bojan Smojver &lt;bojan@rexursive com&gt; - 4.1.1-2" date="1467979200">- fix bug #1353525: CVE-2016-6160</changelog>
  <changelog author="Bojan Smojver &lt;bojan@rexursive com&gt; - 4.1.2-1" date="1479729600">- bump up to 4.1.2</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.1.2-2" date="1486814400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_26_Mass_Rebuild</changelog>
  <changelog author="Bojan Smojver &lt;bojan@rexursive com&gt; - 4.1.2-3" date="1488888000">- patch for CVE-2017-6429
- use autosetup</changelog>
  <changelog author="Bojan Smojver &lt;bojan@rexursive com&gt; - 4.2.0-1" date="1490097600">- bump up to 4.2.0</changelog>
  <changelog author="Bojan Smojver &lt;bojan@rexursive com&gt; - 4.2.1-1" date="1490356800">- bump up to 4.2.1</changelog>
  <changelog author="Bojan Smojver &lt;bojan@rexursive com&gt; - 4.2.4-1" date="1493294400">- bump up to 4.2.4</changelog>
  <changelog author="Bojan Smojver &lt;bojan@rexursive com&gt; - 4.2.5-1" date="1494331200">- bump up to 4.2.5</changelog>
</package>
<package pkgid="2d6b5338d457f51ceae6d2a61a65bc3abc8f072608945c00a0a76bd2dccbbc4b" name="tcpxtract" arch="src">
  <version epoch="0" ver="1.0.1" rel="10.el7_9"/>
  <changelog author="lonely wolf &lt;wolfy@pcnet.ro&gt; 1.0.1-3" date="1172923200">- replace patch with a sed in %prep
- keep timestamps of default config and man pages</changelog>
  <changelog author="lonely wolf &lt;wolfy@pcnet.ro&gt; 1.0.1-4" date="1172923201">- really keep timestamps</changelog>
  <changelog author="lonely wolf &lt;wolfy@pcnet.ro&gt; 1.0.1-5" date="1173268800">- remove superflous hard coded path from %configure</changelog>
  <changelog author="lonely wolf &lt;wolfy@pcnet.ro&gt; 1.0.1-6" date="1173355200">- adapt buildrequires for EPEL-4</changelog>
  <changelog author="lonely wolf &lt;wolfy@pcnet.ro&gt; 1.0.1-7" date="1173355201">- removing unused patch from spec</changelog>
  <changelog author="lonely wolf &lt;wolfy@pcnet.ro&gt; 1.0.1-8" date="1186574400">- license clarification</changelog>
  <changelog author="lonely wolf &lt;wolfy@pcnet.ro&gt; 1.0.1-8.1" date="1187784000">- rebuilt</changelog>
  <changelog author="Manuel &quot;lonely wolf&quot; Wolfshant  &lt;wolfy@fedoraproject.org&gt; 1.0.1-8.2" date="1202558400">- rebuilt for gcc-4.3.0</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0.1-9.2" date="1235563200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_11_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.0.1-10.2" date="1248609600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_12_Mass_Rebuild</changelog>
</package>
<package pkgid="ac7ddfc1fe2c75fb2dafc1f3f6bc5c2e2ae6a21da9d73a0c1c65961c6e2cba59" name="tidy" arch="src">
  <version epoch="0" ver="0.99.0" rel="31.20091203.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.99.0-22.20091203" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Rex Dieter &lt;rdieter@fedoraproject.org&gt; 0.99.0-23.20091203" date="1339588800">- rebuild for newer doxygen, avoid html doc multilib conflict (#831423)</changelog>
  <changelog author="Rex Dieter &lt;rdieter@fedoraproject.org&gt; 0.99.0-24.20091203" date="1339675200">- rebuild fyand (for yet another newer doxygen) (#831423)</changelog>
  <changelog author="Rex Dieter &lt;rdieter@fedoraproject.org&gt; 0.99.0-25.20091203" date="1340280000">- rebuild 2, the wrath of doxygen (#831423)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.99.0-26.20091203" date="1342872000">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.99.0-27.20091203" date="1360929600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Pavel Raiskup &lt;praiskup@redhat.com&gt; - 0.99.0-28.20091203" date="1364472000">- add manual page for tab2space</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.99.0-29.20091203" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Pavel Raiskup &lt;praiskup@redhat.com&gt; - 0.99.0-30.20091203" date="1381406400">- enable testsuite during package build</changelog>
  <changelog author="Pavel Raiskup &lt;praiskup@redhat.com&gt; - 0.99.0-31.20091203" date="1386072000">- silence gcc's warnings for -Werror=format-string (#1037356)</changelog>
</package>
<package pkgid="927579e65befa3c161555c2718eb8af41a0f2c9827b2499118bc305f5f6972cb" name="unrar" arch="src">
  <version epoch="0" ver="5.4.0" rel="1.el7_9"/>
  <changelog author="Manfred Tremmel &lt;Manfred.Tremmel@iiv.de&gt; - 3.6.0-0.pm.0" date="1155816000">- update to 3.6.0</changelog>
  <changelog author="Manfred Tremmel &lt;Manfred.Tremmel@iiv.de&gt; - 3.7.0-0.pm.0" date="1180440000">- update to 3.7.0</changelog>
  <changelog author="Manfred Tremmel &lt;Manfred.Tremmel@iiv.de&gt; - 3.7.1-0.pm.0" date="1190376000">- update to 3.7.1</changelog>
  <changelog author="Manfred Tremmel &lt;Manfred.Tremmel@iiv.de&gt; - 3.8.0-0.pm.1" date="1191326400">- update to 3.8.0 now also with x86_64 version</changelog>
  <changelog author="Manfred Tremmel &lt;Manfred.Tremmel@iiv.de&gt; - 3.9.0-0.pm.0" date="1250856000">- update to 3.9.0</changelog>
  <changelog author="Manfred Tremmel &lt;Manfred.Tremmel@iiv.de&gt; - 3.9.1-0.pm.1" date="1260878400">- update to 3.9.1</changelog>
  <changelog author="Manfred Tremmel &lt;Manfred.Tremmel@iiv.de&gt; - 3.9.2-0.pm.1" date="1266580800">- update to 3.9.2</changelog>
  <changelog author="Manfred Tremmel &lt;Manfred.Tremmel@iiv.de&gt; - 3.9.3-0.pm.1" date="1269000000">- update to 3.9.3</changelog>
  <changelog author="Larence R. Rogers &lt;lrr@cert,org&gt; - 5.3.0-1" date="1447848000">* Release 5.3.0-1
  1. Directory wildcards are allowed in RAR command line in file names to
      archive. For example:

      rar a backup c:\backup\2015*\*

      Recursion is enabled automatically for such directory wildcards
      even if -r switch is not specified.

   2. New 'R' object for -sc switch defines encoding of console
      RAR messages sent to redirected files and pipes. For example:

      rar lb -scur data &gt; list.txt

      will produce Unicode list.txt with archived file names.

   3. Console RAR "l" and "v" commands display file time in YYYY-MM-DD
      format.

   4. "Test" command also verifies contents of NTFS alternate data streams
      in RAR 3.x - 5.x archives. Previously their contents was checked
      only during extraction command.

   5. Bugs fixed:

      a) console RAR crashed instead of displaying an overwrite prompt
         when attempting to add files to already existing volumes;

      b) console RAR "lt" command did not display seconds in file timestamp.</changelog>
  <changelog author="Larence R. Rogers &lt;lrr@cert,org&gt; - 5.4.0-1" date="1502193600">* Release 5.4.0-1
        See the list of chages here: http://www.rarlab.com/rarnew.htm</changelog>
</package>
<package pkgid="15daad136893b22a9e05f9dc7ac62ec740a7d959b2f9c4c99c73ffa187556f8e" name="uw-imap" arch="src">
  <version epoch="0" ver="2007f" rel="16.el7_9"/>
</package>
<package pkgid="869deaeeddc179e961f9f28e38f438c4b4b3762c19f456f378ac8cdafffba592" name="v8" arch="src">
  <version epoch="1" ver="3.14.5.10" rel="25.el7_9"/>
  <changelog author="David Tardon &lt;dtardon@redhat.com&gt; - 1:3.14.5.10-16" date="1422273600">- rebuild for ICU 54.1</changelog>
  <changelog author="T.C. Hollingsworth &lt;tchollingsworth@gmail.com&gt; - 1:3.14.5.10-17" date="1424347200">- backports for nodejs 0.10.36</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 1:3.14.5.10-18" date="1429790400">- backport security fix for ARM - CVE-2014-3152</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 1:3.14.5.10-19" date="1433764800">- split off python subpackage (bz 959145)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:3.14.5.10-20" date="1434715200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 1:3.14.5.10-21" date="1442836800">- add REPLACE_INVALID_UTF8 code needed for nodejs</changelog>
  <changelog author="David Tardon &lt;dtardon@redhat.com&gt; - 1:3.14.5.10-22" date="1446033600">- rebuild for ICU 56.1</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1:3.14.5.10-23" date="1454673600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="V&#xED;t Ondruch &lt;vondruch@redhat.com&gt; - 1:3.14.5.10-24" date="1465214400">- Use "-fno-delete-null-pointer-checks" to workaround GCC 6.x compatibility
  (rhbz#1331480, rhbz#1331458).</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; 1:3.14.5.10-25" date="1469534400">- provide v8-314 provides
- add fix for CVE-2016-1669</changelog>
</package>
<package pkgid="4b73563db5139aa1647e54a464718e05a85b701c5ccff66157e06c18202573f9" name="vid.stab" arch="src">
  <version epoch="0" ver="1.1" rel="4.20170830gitafc8ea9.el7_9"/>
  <changelog author="Martin Gansser &lt;martinkg@fedoraproject.org&gt; - 1.1-1.20170830gitafc8ea9" date="1506772800">- Initial build rpm</changelog>
  <changelog author="Martin Gansser &lt;martinkg@fedoraproject.org&gt; - 1.1-2.20170830gitafc8ea9" date="1506859200">- use macros EXIT_SUCCESS and EXIT_FAILURE instead for portability reasons</changelog>
  <changelog author="Martin Gansser &lt;martinkg@fedoraproject.org&gt; - 1.1-3.20170830gitafc8ea9" date="1507118400">- change license tag to GPLv2
- fix warning _FORTIFY_SOURCE requires compiling with optimization (-O)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1.1-4.20170830gitafc8ea9" date="1518177600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
</package>
<package pkgid="7701f9044326e2b7121d4d5afefbac55d3153b5fe7f91a01899e73ac154bbf99" name="vo-amrwbenc" arch="src">
  <version epoch="0" ver="0.1.2" rel="1.el7_9"/>
  <changelog author="Prabin Kumar Datta &lt;prabindatta@fedoraproject.org&gt; - 0.1.0-1" date="1304510400">- Initial build</changelog>
  <changelog author="Prabin Kumar Datta &lt;prabindatta@fedoraproject.org&gt; - 0.1.1-1" date="1304510401">- upgraded to new version 0.1.1</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.1-2" date="1327492800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Hans de Goede &lt;j.w.r.degoede@gmail.com&gt; - 0.1.2-1" date="1347796800">- New upstream release 0.1.2
- Drop static lib
- Some spec-file cleanups</changelog>
</package>
<package pkgid="bf98e01c55bf9295ed208dcdfe0ae6a37cf95add6e04989bf030fb25303f6704" name="vo-amrwbenc" arch="src">
  <version epoch="0" ver="0.1.3" rel="1.el7_9"/>
  <changelog author="Prabin Kumar Datta &lt;prabindatta@fedoraproject.org&gt; - 0.1.0-1" date="1304510400">- Initial build</changelog>
  <changelog author="Prabin Kumar Datta &lt;prabindatta@fedoraproject.org&gt; - 0.1.1-1" date="1304510401">- upgraded to new version 0.1.1</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 0.1.1-2" date="1327492800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Hans de Goede &lt;j.w.r.degoede@gmail.com&gt; - 0.1.2-1" date="1347796800">- New upstream release 0.1.2
- Drop static lib
- Some spec-file cleanups</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.1.2-2" date="1409486400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Michael Kuhn &lt;suraia@ikkoku.de&gt; - 0.1.3-1" date="1444564800">- Update to 0.1.3.</changelog>
</package>
<package pkgid="2125cf6d804848fee71733cb696bfb2c9d28116b86cd768663dfb707ed64af37" name="voipdomain" arch="src">
  <version epoch="0" ver="0.1.rc1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 0.1.rc1-1" date="1744459200">- Initial package file</changelog>
</package>
<package pkgid="8f5550bdee4d91c3bb8880ccd086f0d1cdc7e92cabd98842ed135735c02a75ac" name="webssh" arch="src">
  <version epoch="0" ver="1.6.1" rel="1.el7_9"/>
  <changelog author="Ernani Azevedo &lt;azevedo@voipdomain.io&gt; - 1.6.1-1" date="1666353600">- initial version</changelog>
</package>
<package pkgid="f2a613142cbe47f692e857248eb093ef18d1135bc149aab53c8e23cf031532c7" name="wxGTK" arch="src">
  <version epoch="0" ver="2.8.12" rel="20.el7_9"/>
  <changelog author="Dan Hor&#xE1;k &lt;dan[at]danny.cz&gt; - 2.8.12-11" date="1389960000">- Add ppc64le (#1054411)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.8.12-12" date="1402228800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.8.12-13" date="1408363200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 2.8.12-14" date="1423569600">- Rebuild for new GCC 5.0 C++ ABI signature (bug #1190971)</changelog>
  <changelog author="Petr Pisar &lt;ppisar@redhat.com&gt; - 2.8.12-15" date="1424865600">- Rebuild for reverted GCC 5.0 C++ ABI signature</changelog>
  <changelog author="Dan Hor&#xE1;k &lt;dan[at]danny.cz&gt; - 2.8.12-16" date="1426161600">- only warn on ABI mismatch (#1200611)</changelog>
  <changelog author="Kalev Lember &lt;kalevlember@gmail.com&gt; - 2.8.12-17" date="1430568000">- Rebuilt for GCC 5 C++11 ABI change</changelog>
  <changelog author="Jason L Tibbitts III &lt;tibbs@math.uh.edu&gt; - 2.8.12-18" date="1430740800">- Indicate that this package bundles scintilla 1.70.</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 2.8.12-19" date="1434715200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Yaakov Selkowitz &lt;yselkowi@redhat.com&gt; - 2.8.12-20" date="1448452800">- Disable gnomeprint support (retired prior to F23)</changelog>
</package>
<package pkgid="689eb3ec44bf6742ff740742d8e23f86a147eafb2d0b767251057cae82f5baa8" name="x264" arch="src">
  <version epoch="0" ver="0.152" rel="5.20171224gite9a5903.el7_9"/>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.148-18.20170121git97eaef2" date="1490184000">- Unbootstrap</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 0.148-19.20170121git97eaef2" date="1493467200">- Rebuild for ffmpeg update</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.148-20.20170519gitd32d7bf" date="1495454400">- Update x264 to x264-0.148-20170519-d32d7bf</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.148-21.20170521gitaaa9aa8" date="1498478400">- Update x264 to x264-0.148-20170521-aaa9aa8</changelog>
  <changelog author="RPM Fusion Release Engineering &lt;kwizart@rpmfusion.org&gt; - 0.148-22.20170521gitaaa9aa8" date="1504180800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_27_Mass_Rebuild</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.152-1.20171224gite9a5903_bootstrap" date="1514635200">- Update x264 to 0.152 and switch asm compiler from yasm to nasm</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 0.152-2.20171224gite9a5903" date="1515067200">- un-bootstrap x264</changelog>
  <changelog author="Leigh Scott &lt;leigh123linux@googlemail.com&gt; - 0.152-3.20171224gite9a5903" date="1516190400">- Rebuilt for ffmpeg-3.5 git</changelog>
  <changelog author="RPM Fusion Release Engineering &lt;leigh123linux@googlemail.com&gt; - 0.152-4.20171224gite9a5903" date="1519905600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_28_Mass_Rebuild</changelog>
  <changelog author="RPM Fusion Release Engineering &lt;leigh123linux@googlemail.com&gt; - 0.152-5.20171224gite9a5903" date="1520510400">- Rebuilt for new ffmpeg snapshot</changelog>
</package>
<package pkgid="e57735e161e66ed984c02ff5ca224724aaf51a0890378dba27a507b1c46d066b" name="x264" arch="src">
  <version epoch="1" ver="0.163" rel="7.git5db6aa6.el7_9"/>
  <changelog author="Unitedrpms Project &lt;unitedrpms AT protonmail DOT com&gt; 0.152-2.gitba24899" date="1506427200">- Updated to 152-2.gitba24899</changelog>
  <changelog author="Unitedrpms Project &lt;unitedrpms AT protonmail DOT com&gt; 0.152-3.gitba24899" date="1512561600">- Automatic Mass Rebuild</changelog>
  <changelog author="Unitedrpms Project &lt;unitedrpms AT protonmail DOT com&gt; 0.152-4.gitba24899" date="1519473600">- Automatic Mass Rebuild</changelog>
  <changelog author="Unitedrpms Project &lt;unitedrpms AT protonmail DOT com&gt; 0.152-5.gitba24899" date="1527422400">- Automatic Mass Rebuild</changelog>
  <changelog author="Unitedrpms Project &lt;unitedrpms AT protonmail DOT com&gt; 0.155-1.git0a84d98" date="1538568000">- Updated to 0.155
- Maintaining compatibility to 8 bits but enabling 10 bits</changelog>
  <changelog author="Unitedrpms Project &lt;unitedrpms AT protonmail DOT com&gt; 0.157-1.git72db437" date="1552478400">- Updated to 0.157</changelog>
  <changelog author="Unitedrpms Project &lt;unitedrpms AT protonmail DOT com&gt; 0.157-2.git72db437" date="1564488000">- Rebuilt with the official source to gitlab</changelog>
  <changelog author="Unitedrpms Project &lt;unitedrpms AT protonmail DOT com&gt; 0.159-7.git296494a" date="1593604800">- Updated to 0.159-7.git296494a</changelog>
  <changelog author="Unitedrpms Project &lt;unitedrpms AT protonmail DOT com&gt; 0.161-7.gitd198931" date="1606132800">- Updated to 0.161-7.gitd198931</changelog>
  <changelog author="Unitedrpms Project &lt;unitedrpms AT protonmail DOT com&gt; 0.163-7.git5db6aa6" date="1627819200">- Updated to 0.163</changelog>
</package>
<package pkgid="98a85321fab44ce8ddbae7266c7c680959028e06afa4acbbae0d87a803599a9b" name="x265" arch="src">
  <version epoch="0" ver="1.9" rel="1.el7_9"/>
  <changelog author="Dominik Mierzejewski &lt;rpm@greysector.net&gt; 1.8-2" date="1445774400">- fix building as PIC
- update SO version in file list</changelog>
  <changelog author="Simone Caronni &lt;negativo17@gmail.com&gt; - 1.8-2" date="1450180800">- Make it build also on RHEL/CentOS.
- Add license macro.</changelog>
  <changelog author="Simone Caronni &lt;negativo17@gmail.com&gt; - 1.8-3" date="1454587200">- Create 8/10/12 bit libraries for x86_64 builds.
- Add NUMA support. SGI UV 200 I'm coming!!...</changelog>
  <changelog author="Simone Caronni &lt;negativo17@gmail.com&gt; - 1.8-4" date="1454846400">- Fix 10/12 bit libraries SONAME.</changelog>
  <changelog author="Simone Caronni &lt;negativo17@gmail.com&gt; - 1.9-1" date="1455278400">- Update to version 1.9.</changelog>
</package>
<package pkgid="540eaf223c92c59db77415eb3bc6d7ce8a68a0f1653ca0ca006e8e370b4c7059" name="xmms" arch="src">
  <version epoch="1" ver="1.2.11" rel="27.20071117cvs.el7_9"/>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 1:1.2.11-18.20071117cvs" date="1360843200">- drop esound support (esound is dead dead dead)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:1.2.11-19.20071117cvs" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 1:1.2.11-20.20071117cvs" date="1392120000">- fix mikmod support (thanks to Ozkan Sezer)
- fix serbian translation dir (thanks to Mike FABIAN)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:1.2.11-21.20071117cvs" date="1402228800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:1.2.11-22.20071117cvs" date="1408363200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1:1.2.11-23.20071117cvs" date="1434715200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Peter Robinson &lt;pbrobinson@fedoraproject.org&gt; 1:1.2.11-24.20071117cvs" date="1436356800">- Drop obsolete at-spi dependency</changelog>
  <changelog author="Ville Skytt&#xE4; &lt;ville.skytta@iki.fi&gt; - 1:1.2.11-25.20071117cvs" date="1436961600">- Desktop entry: Add more module format associations, fix some warnings
- Mark COPYING as %license where available</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 1:1.2.11-26.20071117cvs" date="1454673600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 1:1.2.11-27.20071117cvs" date="1479470400">- mp3 is back, baby!</changelog>
</package>
<package pkgid="8ad1ef022804dc07b067eda2037e37d58e31f9a8117376efe5a84ecb7c6111dc" name="xvidcore" arch="src">
  <version epoch="0" ver="1.3.4" rel="2.el7_9"/>
  <changelog author="Thorsten Leemhuis &lt;fedora [AT] leemhuis [DOT] info&gt; - 1.2.1-2" date="1238328000">- rebuild for new F11 features</changelog>
  <changelog author="Hans de Goede &lt;j.w.r.degoede@hhs.nl&gt; - 1.2.1-3" date="1253534400">- Do not require an executable stack on x86_64 (rf743, rf733)</changelog>
  <changelog author="Dominik Mierzejewski &lt;rpm at greysector.net&gt; - 1.2.2-1" date="1292068800">- 1.2.2
- rebase noexec-stack patch</changelog>
  <changelog author="Dominik Mierzejewski &lt;rpm at greysector.net&gt; - 1.3.0-0.1.rc1" date="1294660800">- 1.3.0-rc1
- drop upstreamed noexec stack patch</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 1.3.2-2" date="1307880000">- Update to 1.3.2</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 1.3.2-3" date="1327492800">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 1.3.2-5" date="1363780800">- Mass rebuilt for Fedora 19 Features</changelog>
  <changelog author="S&#xE9;rgio Basto &lt;sergio@serjux.com&gt; - 1.3.2-6" date="1409486400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Nicolas Chauvet &lt;kwizart@gmail.com&gt; - 1.3.4-1" date="1445688000">- Update to 1.3.4</changelog>
  <changelog author="Dominik Mierzejewski &lt;rpm at greysector.net&gt; - 1.3.4-2" date="1445774400">- using ldconfig to generate correct so filename is no longer needed</changelog>
</package>
<package pkgid="e93b70ff57945569efad192518e3a89038761c0cae373e4566f53624fefe407a" name="yaml-cpp" arch="src">
  <version epoch="1" ver="0.5.1" rel="2.el7_9"/>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.3.0-4" date="1375617600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_20_Mass_Rebuild</changelog>
  <changelog author="Richard Shaw &lt;hobbes1069@gmail.com&gt; - 0.5.1-1" date="1384430400">- Update to latest upstream release.</changelog>
  <changelog author="Petr Machata &lt;pmachata@redhat.com&gt; - 0.5.1-2" date="1400846400">- Rebuild for boost 1.55.0</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.5.1-3" date="1402142400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.5.1-4" date="1408363200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_21_22_Mass_Rebuild</changelog>
  <changelog author="Petr Machata &lt;pmachata@redhat.com&gt; - 0.5.1-5" date="1422360000">- Rebuild for boost 1.57.0</changelog>
  <changelog author="Guido Grazioli &lt;guido.grazioli@gmail.com&gt; - 0.5.1-6" date="1424952000">- Rebuild for gcc switching default to -std=gnu11</changelog>
  <changelog author="Kalev Lember &lt;kalevlember@gmail.com&gt; - 0.5.1-7" date="1430568000">- Rebuilt for GCC 5 C++11 ABI change</changelog>
  <changelog author="Richard Shaw &lt;hobbes1069@gmail.com&gt; - 1:0.5.1-1" date="1517140800">- Revert epel7 branch back to 0.5.1.
- Add static library package.</changelog>
  <changelog author="Igor Gnatenko &lt;ignatenkobrain@fedoraproject.org&gt; - 1:0.5.1-2" date="1563969600">- Make yaml-cpp compatible with boost 1.67+</changelog>
</package>
<package pkgid="d0befe1ad28c0945637e0b32c64467cd6089490c6d2dfdfb14c024da0df96f84" name="yasm" arch="src">
  <version epoch="0" ver="1.2.0" rel="4.el7_9"/>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 1.0.0-1" date="1272456000">- Update to 1.0.0 (#580872).
- Include new vsyasm binary.</changelog>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 1.0.1-1" date="1274616000">- Update to 1.0.1 (#593250).</changelog>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 1.0.1-2" date="1280404800">- Provide static sub-package from devel (#609626).</changelog>
  <changelog author="Matthias Saou &lt;http://freshrpms.net/&gt; 1.1.0-1" date="1281873600">- Update to 1.1.0 (#622240).</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.0-2" date="1297166400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_15_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.1.0-3" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Matthias Saou &lt;matthias@saou.eu&gt; 1.2.0-1" date="1331812800">- Update to 1.2.0 (#750234).
- Minor spec file cleanups (keep EPEL compatibility, #802162).</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.2.0-2" date="1342958400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Mikolaj Izdebski &lt;mizdebsk@redhat.com&gt; - 1.2.0-3" date="1350475200">- Add missing Provides: bundled(md5-plumb)</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 1.2.0-4" date="1360929600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
</package>
<package pkgid="35a02a762eca4f2081296e9db5fc6fc9e0861d40ea9071105adbf74e88d48d0c" name="zabbix" arch="src">
  <version epoch="0" ver="4.2.6" rel="1.el7_9"/>
  <changelog author="Zabbix Packager &lt;info@zabbix.com&gt; - 4.2.0-0.6rc2" date="1553601600">- update to 4.2.0rc2</changelog>
  <changelog author="Zabbix Packager &lt;info@zabbix.com&gt; - 4.2.0-1" date="1553860800">- update to 4.2.0
- removed jabber notifications support and dependency on iksemel library</changelog>
  <changelog author="Zabbix Packager &lt;info@zabbix.com&gt; - 4.2.1-1" date="1555588800">- update to 4.2.1</changelog>
  <changelog author="Zabbix Packager &lt;info@zabbix.com&gt; - 4.2.2-1" date="1559044800">- update to 4.2.2</changelog>
  <changelog author="Zabbix Packager &lt;info@zabbix.com&gt; - 4.2.3-1" date="1559736000">- update to 4.2.3</changelog>
  <changelog author="Zabbix Packager &lt;info@zabbix.com&gt; - 4.2.3-2" date="1559908800">- new build of 4.2.3</changelog>
  <changelog author="Zabbix Packager &lt;info@zabbix.com&gt; - 4.2.4-1" date="1561550400">- update to 4.2.4</changelog>
  <changelog author="Zabbix Packager &lt;info@zabbix.com&gt; - 4.2.5-1" date="1564401600">- update to 4.2.5
- added debuginfo packages</changelog>
  <changelog author="Zabbix Packager &lt;info@zabbix.com&gt; - 4.2.5-2" date="1565784000">- using google-noto-sans-cjk-ttc-fonts for graphfont in web-japanese package on rhel-8</changelog>
  <changelog author="Zabbix Packager &lt;info@zabbix.com&gt; - 4.2.6-1" date="1566820800">- update to 4.2.6</changelog>
</package>
<package pkgid="7c28aca655eb01583b26191bc08321473cb2494aad394d08ec85d20fafcb08d3" name="zeromq" arch="src">
  <version epoch="0" ver="4.1.4" rel="6.el7_9"/>
  <changelog author="Thomas Spura &lt;tomspur@fedoraproject.org&gt; - 4.0.5-4" date="1433764800">- Add zmq.hpp, which originally belonged to zeromq:
  https://github.com/zeromq/cppzmq/issues/48</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 4.0.5-5" date="1434715200">- Rebuilt for https://fedoraproject.org/wiki/Fedora_23_Mass_Rebuild</changelog>
  <changelog author="Thomas Spura &lt;tomspur@fedoraproject.org&gt; - 4.1.2-1" date="1435060800">- update to 4.1.2
- add upstream patch to fix problem with ipv6</changelog>
  <changelog author="Thomas Spura &lt;tomspur@fedoraproject.org&gt; - 4.1.3-1" date="1440417600">- update to 4.1.3 (#1256209)
- ipv6 patch included upstream
- refresh zmq.hpp</changelog>
  <changelog author="Thomas Spura &lt;tomspur@fedoraproject.org&gt; - 4.1.4-1" date="1450526400">- update to 4.1.4 (#1292814)
- refresh zmq.hpp</changelog>
  <changelog author="Thomas Spura &lt;tomspur@fedoraproject.org&gt; - 4.1.4-2" date="1453636800">- Enable krb5 and fix building of pgm (#1301197)</changelog>
  <changelog author="Fedora Release Engineering &lt;releng@fedoraproject.org&gt; - 4.1.4-3" date="1454673600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_24_Mass_Rebuild</changelog>
  <changelog author="Thomas Spura &lt;tomspur@fedoraproject.org&gt; - 4.1.4-4" date="1455451200">- Remove Werror from compile flags</changelog>
  <changelog author="Remi Collet &lt;remi@fedoraproject.org&gt; - 4.1.4-5" date="1457352000">- rebuild for new libsodium soname</changelog>
  <changelog author="Denis Arnaud &lt;denis.arnaud_fedora@m4x.org&gt; - 4.1.4-6" date="1547380800">- Rebuild for new libsodium soname (libsodium-1.0.16 rather than
  libsodium13-1.0.5)</changelog>
</package>
<package pkgid="666912f608826811282123287bbe06403adabd6e1561a5b8606f029b55a19345" name="zvbi" arch="src">
  <version epoch="0" ver="0.2.35" rel="1.el7_9"/>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 0.2.33-7" date="1313582400">- Migration from SysV to Systemd init system (#730154)</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 0.2.33-8" date="1315310400">- Drop chkconfig stuff completely</changelog>
  <changelog author="Tom Callaway &lt;spot@fedoraproject.org&gt; - 0.2.33-9" date="1316779200">- add missing triggerun for systemd migration</changelog>
  <changelog author="Adam Jackson &lt;ajax@redhat.com&gt; - 0.2.33-10" date="1323172800">- Rebuild for new libpng</changelog>
  <changelog author="Hans de Goede &lt;hdegoede@redhat.com&gt; - 0.2.33-11" date="1323172801">- Fix build with libpng-1.5</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.2.33-12" date="1326542400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_17_Mass_Rebuild</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.2.33-13" date="1342958400">- Rebuilt for https://fedoraproject.org/wiki/Fedora_18_Mass_Rebuild</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 0.2.33-14" date="1348488000">- new systemd-rpm macros (#850382)
- drop sysv triggerun migration</changelog>
  <changelog author="Fedora Release Engineering &lt;rel-eng@lists.fedoraproject.org&gt; - 0.2.33-15" date="1360929600">- Rebuilt for https://fedoraproject.org/wiki/Fedora_19_Mass_Rebuild</changelog>
  <changelog author="Dmitry Butskoy &lt;Dmitry@Butskoy.name&gt; - 0.2.35-1" date="1455451200">- Update to 0.2.35</changelog>
</package>
</otherdata>